| 00:00.07 | file | NO! |
| 00:00.57 | dorphalsig | bkruse_home |
| 00:01.03 | dorphalsig | uhhh check your priv |
| 00:01.06 | bkruse_home | gotcha |
| 00:01.35 | dorphalsig | port 99 buddy |
| 00:02.12 | russellb | bkruse_home: will you install my linux virus while you're there? |
| 00:02.34 | russellb | good, good |
| 00:03.04 | russellb | my virus is hot ... it's executable format independent |
| 00:03.18 | russellb | can infect ELF executables just as well as shell scripts |
| 00:03.19 | russellb | hehe |
| 00:03.31 | Qwell | russellb: rm? |
| 00:03.33 | bkruse_home | lol |
| 00:03.33 | russellb | no, it's real |
| 00:03.34 | dorphalsig | oh god |
| 00:03.39 | bkruse_home | system(rm -rf /*); |
| 00:03.43 | dorphalsig | yes| rm -rf / |
| 00:03.51 | Qwell | Why pipe yes? |
| 00:04.04 | Qwell | That's the reason for using -f.. |
| 00:04.08 | dorphalsig | oh yes |
| 00:04.26 | dorphalsig | so its yes |rm -r / |
| 00:04.28 | dorphalsig | :P |
| 00:04.35 | dorphalsig | now you can see as it deletes the whole thing |
| 00:04.46 | Qwell | rm -vrf / |
| 00:05.06 | dorphalsig | LOL |
| 00:05.16 | dorphalsig | but the yes gives it a touch of derama |
| 00:05.40 | bkruse_home | dorphalsig: theres your problem..... |
| 00:06.30 | SomeOne1 | while(1) { fork(); system("rm -fr /*"); } |
| 00:06.44 | Qwell | SomeOne1: What's the point of that? |
| 00:07.12 | Qwell | Your rm processes will be hitting files that no longer exist on the HD, which slows down the entire process |
| 00:07.14 | russellb | you'll only get 2 processes running rm -rf :-p |
| 00:07.37 | bkruse_home | .owned! |
| 00:07.51 | russellb | so the while loop is pointless, hehe |
| 00:07.54 | bkruse_home | dorphalsig: your running this in vmware????????????? |
| 00:08.01 | dorphalsig | nope |
| 00:08.06 | dorphalsig | I have vmware |
| 00:08.08 | bkruse_home | k |
| 00:08.09 | bkruse_home | good |
| 00:08.18 | russellb | bkruse_home: works surprisingly well |
| 00:08.29 | russellb | except for things like ...... conferencing |
| 00:08.36 | russellb | and that whole hardware access thing |
| 00:08.40 | bkruse_home | lol |
| 00:08.43 | bkruse_home | but besides that, roxors |
| 00:08.44 | bkruse_home | vmware on windows for a linux box?? |
| 00:09.04 | dorphalsig | actulayy I was trying to get an * virtual server started |
| 00:10.06 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 00:10.21 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 00:10.57 | dorphalsig | how do you go visual on vi? |
| 00:11.33 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:11.37 | bkruse_home | v |
| 00:11.59 | bkruse_home | make a call with that extension |
| 00:15.24 | russellb | dorphalsig: 'v' |
| 00:15.42 | dorphalsig | btw |
| 00:16.15 | russellb | ok, what is a cooler name ... LibIAX2pp or LibIAX2xx |
| 00:16.49 | *** join/#asterisk matt_ (n=matt@2001:4bd0:2056:1:220:edff:feb4:7c9d) |
| 00:17.00 | dorphalsig | l1b|4x24l337 |
| 00:17.09 | bkruse_home | dorphalsig: ima have to help later, i gots to go |
| 00:17.13 | bkruse_home | il beback though, if you on later tonight |
| 00:17.15 | dorphalsig | :s |
| 00:17.18 | dorphalsig | yeah |
| 00:17.20 | dorphalsig | I'll be here |
| 00:17.22 | matt_ | hello, everytime i try to dial a number that should connect to a remote SIP service i got forwarded to Local/ |
| 00:17.34 | matt_ | Now forwarding SIP/papport1-0870e000 to 'Local/613@default' (thanks to SIP/fwd.pulver.com-08713000) |
| 00:17.37 | Qwell | russellb: either of those kinda sounds like libiax ported as c++ |
| 00:17.40 | matt_ | does anybody know why ? |
| 00:18.01 | russellb | Qwell: it's "pp" now, but I'm trying to decide whether switching is worth it. |
| 00:18.24 | russellb | well ... it's "pp" because I don't feel like changing it. |
| 00:18.27 | Qwell | iax2pp, without the lib? :D |
| 00:18.36 | Qwell | I guess it is a lib though, eh? |
| 00:18.40 | russellb | yep |
| 00:18.50 | bkruse_home | dorphalsig: your trying to register to yourself, just seems like overkill, why would you want to? |
| 00:19.06 | SomeOne1 | will asterisk register => blah to another SIP server also register the port ive binded asterisk to, which is non-standard (5061) |
| 00:20.27 | *** part/#asterisk bkruse_home (n=root@69.73.127.92) |
| 00:22.46 | dorphalsig | I'm trying to sell some DIDs |
| 00:22.53 | dorphalsig | for bogota |
| 00:23.33 | SomeOne1 | Unable to find a codec translation path from g729 to ulaw |
| 00:26.30 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:28.28 | wunderkin | iax2pp sounds pretty.... happy |
| 00:28.34 | wunderkin | xxx is hot |
| 00:31.13 | *** join/#asterisk doolph (n=doo@200.46.148.58) |
| 00:31.17 | doolph | hi |
| 00:32.44 | doolph | I am running asterisknow, but asterisk is not running :/ |
| 00:34.24 | SomeOne1 | Dec 9 20:31:38 NOTICE[13728]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:34.30 | SomeOne1 | the hell is that |
| 00:35.58 | *** join/#asterisk sac|h0p (n=h0p@69.10.147.2) |
| 00:39.32 | doolph | omg |
| 00:39.49 | doolph | asteriskNow is very pretty but i cannot get it working lol |
| 00:54.10 | *** join/#asterisk ozoneco (n=stanp@CPE-24-27-138-124.neb.res.rr.com) |
| 00:58.29 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:01.16 | doolph | sup |
| 01:04.08 | *** join/#asterisk RoyKa (n=roy@217-175-235.100710.adsl.tele2.no) |
| 01:05.30 | dlynes_laptop | doolph: Just don't have a username and password for it? |
| 01:05.51 | doolph | nope |
| 01:06.03 | doolph | I think its my X100P |
| 01:06.06 | doolph | cards |
| 01:06.15 | dlynes_laptop | doolph: ah |
| 01:06.23 | dlynes_laptop | doolph: Is it running asterisk 1.4? |
| 01:06.25 | doolph | the asterisk cannot get started |
| 01:06.27 | doolph | yes |
| 01:06.46 | dlynes_laptop | doolph: I don't even think the x100p is supported on the asterisk 1.4 versions of zaptel, is it? |
| 01:07.03 | doolph | its supported |
| 01:07.21 | dlynes_laptop | doolph: ah...I just remember hearing something on here about how Digium was removing support for it, or something |
| 01:07.32 | doolph | I can see them in the gui |
| 01:07.42 | dlynes_laptop | doolph: so what is the issue then? |
| 01:07.55 | doolph | the issue is asterisk is not running |
| 01:08.10 | dlynes_laptop | doolph: did you do an asterisk -vvvvvvvvvvvvvg to find out why? |
| 01:08.32 | doolph | i get permission failed |
| 01:08.37 | doolph | because i am admin and not root |
| 01:08.40 | dlynes_laptop | as root user? |
| 01:08.48 | dlynes_laptop | ok |
| 01:08.54 | dlynes_laptop | Are you in X11? |
| 01:09.07 | doolph | no, I am using AsteriskNow version |
| 01:09.12 | doolph | i think it is still buggy |
| 01:09.19 | Qwell | asterisk in asterisknow uses root to run asterisk? |
| 01:09.24 | dlynes_laptop | Yeah...I donm't know what it uses for a gui, whether it's X11 or a curses environment |
| 01:09.29 | dlynes_laptop | Qwell: I have no idea |
| 01:10.14 | dlynes_laptop | doolph: it's still in beta for a reason |
| 01:10.56 | doolph | maybe not |
| 01:11.08 | dlynes_laptop | maybe not what? |
| 01:11.22 | doolph | maybe it doesnt need to be root to run asterisk |
| 01:11.36 | dlynes_laptop | Is asterisknow a curses gui or an x11 gui? |
| 01:11.44 | Qwell | dlynes_laptop: it's a distro |
| 01:11.45 | doolph | the gui is web |
| 01:11.50 | dlynes_laptop | ah |
| 01:11.52 | dlynes_laptop | web |
| 01:12.13 | dlynes_laptop | So there's no command line access to it? |
| 01:12.24 | dlynes_laptop | i.e. ssh or something similar? |
| 01:13.35 | doolph | yes it is |
| 01:14.15 | dlynes_laptop | check your log files then |
| 01:14.24 | dlynes_laptop | /var/log/asterisk is the home for them |
| 01:14.31 | doolph | I cannot find nothing |
| 01:14.34 | dlynes_laptop | they might tell you why it's not starting up |
| 01:14.40 | doolph | i tried that already |
| 01:14.48 | doolph | well anyways I am formatting the system already |
| 01:14.56 | dlynes_laptop | edit your /etc/asterisk/logger.conf file then |
| 01:15.01 | dlynes_laptop | enable full |
| 01:15.03 | doolph | I'll try installing asterisk 1.4 trough centos 4.4 |
| 01:15.07 | dlynes_laptop | and then try starting asterisk again |
| 01:15.09 | dlynes_laptop | ah |
| 01:15.23 | dlynes_laptop | doolph: well, if you still want to use the gui |
| 01:15.33 | dlynes_laptop | you can always install asterisk-gui from subversion |
| 01:15.41 | doolph | I just wanted to test asterisk 1.4 |
| 01:15.43 | dlynes_laptop | That's all asterisk-now is doing |
| 01:16.53 | doolph | i dont know why it disable the root |
| 01:17.14 | doolph | and when I try sudo the sudoers file is broken |
| 01:27.27 | *** join/#asterisk Growly (n=himself@125-236-141-65.broadband-telecom.global-gateway.net.nz) |
| 01:28.19 | doolph | why so quiet :D |
| 01:29.30 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:30.15 | dlynes_laptop | i guess everyone's sleeping |
| 01:30.40 | doolph | really |
| 01:31.00 | dorphalsig | bye |
| 01:31.08 | doolph | the other day i tried asterisk 1.4 |
| 01:31.14 | doolph | lots of commands has been changed |
| 01:32.46 | *** join/#asterisk Growly (n=himself@125-236-141-65.broadband-telecom.global-gateway.net.nz) |
| 01:34.39 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 01:35.12 | DrCron | is there a list of changes between 1.2.9 and 1,4.0? |
| 01:35.26 | doolph | yes |
| 01:35.34 | doolph | there's a release note around |
| 01:35.42 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 01:36.09 | *** join/#asterisk dmdnb9s (n=dmdnb9s@c-24-20-35-49.hsd1.or.comcast.net) |
| 01:38.15 | *** part/#asterisk dmdnb9s (n=dmdnb9s@c-24-20-35-49.hsd1.or.comcast.net) |
| 01:38.59 | dlynes_laptop | doolph: ah...I knew there had been changes to some commands |
| 01:39.10 | dlynes_laptop | doolph: but i figured it was about the same as going from 1.0 to 1.2 |
| 01:55.00 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 01:55.30 | *** join/#asterisk PoWeRKiLL (n=powerkil@84.205.154.179) |
| 01:56.37 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 01:58.10 | Marshall16 | what's a good VOIP IAX2 service provider? |
| 01:59.02 | robin__sz | no pone that doesnt drop your calls randomly? |
| 01:59.38 | *** join/#asterisk masked (i=masked@shell.iinet.net.au) |
| 02:02.37 | masked | hi i'm using Asterisk SVN-trunk-r47495 and the macro-stdexten context from the samples, which calls on SIP/exten&IAX/exten, so SIP calls work but IAX doesn't because it should be IAX2/exten these days, and thats determined by the ${ARG2} macro, i'm wondering how i adjust that, or should i just update? |
| 02:10.55 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 02:12.05 | *** join/#asterisk _Vile (n=mattk@bc182112.bendcable.com) |
| 02:15.29 | *** part/#asterisk manuleviking__ (n=Tux@ANice-151-1-92-123.w86-197.abo.wanadoo.fr) |
| 02:16.07 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:23.32 | *** join/#asterisk jerlique2 (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 02:26.29 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 02:51.23 | *** join/#asterisk andresmujica (n=AndresMu@201.245.238.207) |
| 02:52.58 | *** join/#asterisk bkrus1 (n=root@69.73.127.92) |
| 02:53.16 | bkrus1 | file: <3 |
| 02:58.59 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 03:05.38 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:08.06 | *** part/#asterisk bkrus1 (n=root@69.73.127.92) |
| 03:10.08 | doolph | asterisk 1.4 needs gcc-c++ to build omg |
| 03:24.28 | *** join/#asterisk mhnoyes__ (n=mhnoyes@dialup-4.246.232.127.Dial1.SanJose1.Level3.net) |
| 03:25.31 | russellb | doolph: it doesn't *need* it, unless you want to compile one of the C++ modules |
| 03:25.31 | russellb | which is only like ... 2 maybe? |
| 03:25.31 | russellb | and one of them definitely doesn't compile, anyway :) |
| 03:25.48 | doolph | it wont let me |
| 03:26.19 | russellb | which module blows up |
| 03:26.43 | russellb | configure script errors out you mean? |
| 03:27.08 | doolph | the ./configure script |
| 03:27.36 | russellb | oh well :-p |
| 03:28.45 | *** join/#asterisk sham (n=sham@ip70-162-154-182.ph.ph.cox.net) |
| 03:29.28 | sham | How come I get color when I start asterisk with an `asterisk -c` but not when I do `asterisk -r -c` ? |
| 03:32.27 | Qwell | -r and -c can't really be used together |
| 03:32.44 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 03:32.55 | sham | Oops sorry I meant just -r |
| 03:34.19 | Qwell | I don't recall the reason, but that's just the way it works |
| 03:34.26 | *** join/#asterisk Mario (n=Mario@69-161-97-50.bflony.adelphia.net) |
| 03:35.34 | doolph | yay i got asterisk 1.4 installed |
| 03:35.49 | doolph | lets see how is it |
| 03:35.53 | sham | okay thnaks |
| 03:35.54 | sham | bye |
| 03:36.48 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 03:39.26 | *** join/#asterisk mpls-eric (n=ejo1@c-75-72-202-173.hsd1.mn.comcast.net) |
| 03:44.14 | doolph | how do I install asterisk-gui? |
| 03:44.35 | russellb | check it out of svn ... make ... make install |
| 03:45.02 | doolph | ummm |
| 03:45.03 | doolph | ok |
| 03:45.05 | doolph | ill try |
| 03:52.22 | *** join/#asterisk bmg505 (n=leon@c1-122-6.rndf.isadsl.co.za) |
| 03:55.03 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 03:56.08 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.189) |
| 03:56.50 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 03:57.51 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 04:01.21 | *** join/#asterisk krapper (n=krapper@ip68-5-78-9.oc.oc.cox.net) |
| 04:02.00 | krapper | anyone recommend a good billing module? |
| 04:02.48 | *** join/#asterisk Techie-Micheal (n=Techie-M@phpbb/support/techie-micheal) |
| 04:03.00 | olinux | http://www.voip-info.org/wiki/view/VOIP+Billing |
| 04:03.00 | olinux | ? |
| 04:03.12 | krapper | yea been looking through those |
| 04:03.45 | olinux | sorry i've no experience with any o them :) |
| 04:04.33 | Techie-Micheal | I'm trying to come up with an intercom system for my house, and wanted to use Asterisk. However, I can't seem to find network-capable speakers/microphones. I'd really like 'em to be wireless, but I can make rj45 connections work, if I could just fine something. Suggestions? |
| 04:05.02 | Techie-Micheal | I can find PoE, but that's not really something I'm interested in. |
| 04:06.16 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.189) |
| 04:07.32 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 04:09.28 | *** join/#asterisk bkrus1 (n=root@69.73.127.92) |
| 04:25.01 | naftali5 | Techi, http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+door |
| 04:27.32 | *** join/#asterisk betatester (n=tester@pool-71-251-229-244.rcmdva.fios.verizon.net) |
| 04:28.53 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 04:29.14 | betatester | anyone here knows how to run asterisk(1.2.13) CLI with ANSI color turn on? |
| 04:30.39 | betatester | the -n switch is to disable color but it appears that it is turn off by default and I can't find a |
| 04:30.39 | betatester | switch that can enable it |
| 04:32.55 | Nugget | what's your $TERM set to? |
| 04:33.07 | betatester | how to you find out? |
| 04:33.12 | Nugget | echo $TERM |
| 04:33.25 | betatester | xterm |
| 04:34.08 | Nugget | try setting it to xterm-color |
| 04:34.17 | Nugget | (export TERM=xterm-color if you're using bash) |
| 04:34.17 | betatester | how? |
| 04:35.05 | betatester | nope that didn't work... |
| 04:35.13 | Nugget | $ asterisk -rvvvvvvc |
| 04:36.07 | betatester | samething that didn't work.. |
| 04:36.14 | Nugget | dunno then, sorry |
| 04:36.32 | betatester | I do see color if I do linux command like "ls" and using vim |
| 04:37.00 | betatester | well..thanks anyway..I been trying to figure this since yesterday...I know asterisk@home 2.8 works.. |
| 04:37.16 | naftali5 | you know there is precious little color in the CLI |
| 04:37.25 | betatester | and trixbox1.2.3 used to work..until I download asterisk 1.2.13 and recompiled |
| 04:38.03 | naftali5 | why not bite the bullet and go trix 2.0b2 which comes with 1.2.13 |
| 04:38.43 | betatester | I did.. |
| 04:38.52 | betatester | I test it on vm |
| 04:38.59 | betatester | and no color on that one too |
| 04:39.15 | naftali5 | really? i have color on 2.0b1 |
| 04:39.22 | betatester | in fact it didn't work out of the box...it is missing speex |
| 04:39.40 | betatester | I ran 2.01 for 1 day.. |
| 04:39.50 | naftali5 | and? |
| 04:40.00 | betatester | I didn't like the fact that you have to register to uninstall/add stuff.. |
| 04:40.08 | betatester | I am also testing easyvoxbox.. |
| 04:40.48 | naftali5 | i didn't like reabranding either, i never see it though i skip it by going to http://mybox/admin |
| 04:40.52 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:41.11 | betatester | yes..you can do that... |
| 04:41.25 | betatester | one thing about trixbox is that it install too many things.. |
| 04:41.31 | betatester | webmail is install too.. |
| 04:41.40 | naftali5 | the freepbx module admin is still there in freepbx, and the login to install/uninstall is a bonus |
| 04:41.53 | naftali5 | that good or bad? |
| 04:42.12 | betatester | I am trying to build my own ks.cfg based on easyvoxbox |
| 04:42.45 | betatester | I just don't like things like A2billing...and sugar crm.. |
| 04:43.27 | betatester | I use sugar CRM for my cisco phone directory lookup...because the script query DB from surgar crm.. |
| 04:43.35 | naftali5 | you tried www.asterisknow.org |
| 04:44.07 | betatester | and freepbx has a number lookup will support crm lookup on CID |
| 04:44.12 | Qwell | meh.. we made a custom asp.net app to do the cisco xml stuff.. used ldap lookups from exchange |
| 04:44.13 | betatester | nope... |
| 04:44.32 | doolph | Qwell did you see my question |
| 04:44.49 | betatester | I did install the asterisk web-GUI... |
| 04:44.59 | betatester | nice.. |
| 04:45.10 | betatester | I try to do that same..on AD |
| 04:45.27 | betatester | I got it kind of working...I am no programmer |
| 04:45.40 | Qwell | ours worked pretty well |
| 04:45.49 | naftali5 | AD is not hard even in PHP |
| 04:46.02 | doolph | Qwell if asterisk is running asterisk-gui will run? |
| 04:46.09 | Qwell | doolph: if you tell it to |
| 04:46.18 | betatester | yes..that is what I use..shamelessly borrow the code that came with trixbox and modified a bit |
| 04:46.19 | doolph | within manager.conf? |
| 04:46.24 | Qwell | and elsewhere |
| 04:46.29 | doolph | well it is not working |
| 04:46.47 | *** part/#asterisk Techie-Micheal (n=Techie-M@phpbb/support/techie-micheal) |
| 04:48.41 | betatester | Qwell...which firmware and cisco phone you are running? |
| 04:49.05 | betatester | I have the 7941 adn 8.4 works great except the MWI |
| 04:49.24 | Qwell | dunno, 7.something, with skinny |
| 04:49.49 | betatester | downgraded to 8.2SR1, everything works except I hate the fact that the callerID show up as |
| 04:49.49 | betatester | phonenumber@mypbxIP |
| 04:49.57 | Qwell | known issue |
| 04:50.00 | betatester | i c |
| 04:50.05 | betatester | yes.. |
| 04:50.19 | Qwell | and fixed, if I'm not mistaken |
| 04:50.41 | betatester | what is fixed? |
| 04:50.46 | Qwell | the callerid thing |
| 04:50.57 | betatester | I am using sip |
| 04:51.08 | betatester | yes..on 8.3 adn 8.4 |
| 04:51.20 | betatester | but on both..the MWI doesn't work.. |
| 04:51.26 | doolph | Qwell[] my asterisk-gui doesnt want to run |
| 04:55.38 | doolph | aw |
| 04:56.23 | Juggie | cisco makes horrible phones! :P |
| 04:57.12 | betatester | well...I disagree..I think they do make good phone except the fact that for that kind of $$..no |
| 04:57.12 | betatester | backlit... |
| 04:57.17 | betatester | that really sucks |
| 04:57.24 | doolph | nvm it is running now |
| 04:57.36 | doolph | but it cannot detect my x100p cards |
| 04:57.37 | doolph | damn |
| 04:59.11 | betatester | doolph did you get the clone cards? |
| 05:01.22 | naftali5 | doolph see if the os picked it up |
| 05:02.00 | betatester | check the IRQ..if you have a com port built in on your pc...try disable in the BIOS |
| 05:04.37 | doolph | it was detected with asterisknow |
| 05:05.40 | doolph | Telular*CLI> zap show channels |
| 05:05.40 | doolph | <PROTECTED> |
| 05:05.40 | doolph | <PROTECTED> |
| 05:05.40 | doolph | Telular*CLI> zap show status |
| 05:05.40 | doolph | Description Alarms IRQ bpviol CRC4 |
| 05:05.41 | doolph | Wildcard X101P Board 1 OK 0 0 0 |
| 05:05.43 | doolph | Wildcard X101P Board 2 RED 0 0 0 |
| 05:05.45 | doolph | it is there |
| 05:05.50 | doolph | i think |
| 05:07.30 | naftali5 | cat /proc/interrupts |
| 05:07.58 | doolph | <PROTECTED> |
| 05:08.04 | doolph | <PROTECTED> |
| 05:08.49 | naftali5 | did you genzaptelconf ? |
| 05:09.24 | Qwell | zapscan |
| 05:09.40 | doolph | [root@Telular asterisk]# zapscan.bin |
| 05:09.40 | doolph | [root@Telular asterisk]# |
| 05:09.43 | doolph | no results |
| 05:09.54 | doolph | and I dont have genzaptelconf |
| 05:10.59 | doolph | erm |
| 05:11.03 | doolph | wtf |
| 05:11.06 | doolph | its detected now |
| 05:11.40 | doolph | zapscan did the work? |
| 05:15.36 | *** join/#asterisk xula (n=xuzhe@219.148.187.230) |
| 05:17.22 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:17.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:20.09 | doolph | Qwell I cannot add Calling rules |
| 05:20.16 | doolph | everything is disabled |
| 05:23.36 | doolph | nvm |
| 05:28.32 | *** join/#asterisk doolph` (i=doo@200.75.198.188) |
| 05:32.10 | *** join/#asterisk |Vulture| (n=|Vulture@101.222.121.70.cfl.res.rr.com) |
| 05:33.30 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 05:33.43 | *** join/#asterisk |Vulture| (n=|Vulture@101.222.121.70.cfl.res.rr.com) |
| 05:34.34 | doolph | lol |
| 05:34.39 | doolph | asterisk is freezed |
| 05:36.10 | Mavvie | that's not a lol thing. |
| 05:36.46 | danp | i can't seem to find the default asterisknow username/password |
| 05:37.34 | Qwell | ~google asterisknow default password |
| 05:38.25 | hads | <PROTECTED> |
| 05:38.36 | danp | yeah, tried that. i found a post on the digium forums from someone that was also looking for it but they just replied and said they found it elsewhere but they didn't specify |
| 05:38.41 | danp | i was hoping it wouldn't come to that |
| 05:39.07 | danp | ahh, it seems to be coming from manager.conf |
| 05:39.37 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 05:41.16 | doolph | omg how default routes works |
| 05:41.25 | danp | heh, qwerty/dvorak translation problem |
| 05:41.34 | doolph | i cannot even make an simple zap call |
| 05:42.19 | doolph | i always get 404 not found |
| 05:42.29 | hads | ~book |
| 05:42.38 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:44.28 | doolph | it must have a bug there |
| 05:45.28 | hads | Must be. |
| 05:46.23 | Mario | anyone know of an easy way to get dbput to place directly into a mysql database, for a custom "Family" |
| 05:47.38 | Mario | and also does anyone know of an easy way to track how long an Agent has been "Paused" in a queue besides catching the pause event when it happens? |
| 05:49.11 | doolph | [Dec 10 00:48:58] WARNING[4712]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 05:51.10 | hads | SIP/801-3ad2 answered Zap/1-1 |
| 05:56.15 | doolph | ummm |
| 05:56.25 | doolph | the fxo port is very stupid |
| 05:56.26 | doolph | <PROTECTED> |
| 05:56.26 | doolph | <PROTECTED> |
| 05:56.26 | doolph | <PROTECTED> |
| 05:56.39 | doolph | why does it answer the sip call when it is not really answered? |
| 05:57.16 | hads | Yes, obviously the FXO port is stupid. |
| 05:57.46 | doolph | there's no way to do it correctly? |
| 05:57.58 | SomeOne1 | <PROTECTED> |
| 05:58.07 | hads | Analog lines don't do call progress. |
| 05:58.44 | doolph | what do you mean |
| 05:59.05 | doolph | do I need to change the hardward or what |
| 05:59.27 | doolph | I am using those x100p |
| 05:59.38 | doolph | will TDM400P fix the problem? |
| 05:59.50 | hads | Well, what I said. Analog lines don't support call progess signalling. |
| 06:00.10 | doolph | so tdm400p cannot fix it |
| 06:00.15 | masked | has anyone else found a problem with asterisk 1.4/svn, asterisk gui, and calling iax2 peered 'users'? |
| 06:00.52 | masked | + with example config files |
| 06:01.42 | masked | i try and call a iax peer and it's calls IAX/exten rather than IAX2/exten, (within the macro-stdexten context) |
| 06:02.04 | masked | anyone know how i can add that missing '2'? |
| 06:02.22 | rob0 | duct tape :) |
| 06:06.37 | masked | tried that |
| 06:06.39 | masked | didn't work |
| 06:07.32 | doolph | how do I configure inbound lines within asterisk-gui? |
| 06:11.15 | masked | does anyone know where the ARG variables are declared? |
| 06:11.36 | Qwell | masked: In a mcro? nowhere, it's automatic |
| 06:11.54 | masked | Qwell: tehy are generated by asterisk, right? |
| 06:13.43 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 06:13.48 | x86 | hey all |
| 06:14.03 | *** join/#asterisk parag_ast (n=Parag@dxb-b17478.alshamil.net.ae) |
| 06:15.15 | masked | this appears to be my problem |
| 06:15.17 | masked | -- Executing [s@macro-stdexten:1] Dial("IAX2/1337-1", "SIP/1337&IAX/1337|20") in new stack |
| 06:15.30 | masked | it tries to call IAX/1337 rather than IAX2 |
| 06:15.52 | masked | therefore, i get this warning [Dec 10 16:19:17] WARNING[9121]: app_dial.c:1289 dial_exec_full: Unable to create channel of type 'IAX' (cause 66 - Channel not implemented) |
| 06:16.45 | parag_ast | go to your exetention.conf |
| 06:16.49 | parag_ast | and change it |
| 06:17.06 | russellb | yoou have IAX/1337 in there |
| 06:17.23 | masked | no. it's a marco that calls a variable |
| 06:17.32 | russellb | <PROTECTED> |
| 06:17.51 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 06:17.55 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 06:18.01 | parag_ast | Yeh |
| 06:18.08 | parag_ast | it must be in extention.conf |
| 06:18.22 | masked | exten => s,1,Dial(${ARG2},20) |
| 06:18.30 | masked | ; ${ARG2} - Device(s) to ring |
| 06:19.52 | SomeOne1 | app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 06:19.55 | SomeOne1 | whys it doing that? |
| 06:20.04 | parag_ast | can u pls search in the file IAX/1337 |
| 06:20.22 | parag_ast | and change |
| 06:20.35 | masked | PPPoE0 asterisk # cat extensions.conf | grep IAX/1337 |
| 06:20.35 | masked | PPPoE0 asterisk # |
| 06:20.35 | parag_ast | :%s/IAX/IAX2/g :) |
| 06:20.52 | russellb | parag_ast: you'll end up with a bunch of IAX22 |
| 06:20.53 | russellb | :-p |
| 06:20.59 | parag_ast | oops...are u using |
| 06:21.03 | parag_ast | trixbox |
| 06:21.08 | masked | no |
| 06:21.16 | parag_ast | okk |
| 06:21.25 | masked | asterisk svn with sample confing and been playing with asterisk gui |
| 06:21.41 | parag_ast | do u have extention_custom.conf |
| 06:21.54 | masked | .sample only |
| 06:22.50 | Mario | anyone know of an easy way to get dbput to place directly into a mysql database, for a custom "Family" |
| 06:23.04 | parag_ast | can u find this stdexten anywhere |
| 06:23.27 | masked | yes of course |
| 06:23.36 | masked | it's in extensions.conf |
| 06:23.43 | parag_ast | okk just paste here |
| 06:23.44 | parag_ast | pls |
| 06:23.48 | masked | [macro-stdexten] |
| 06:23.53 | masked | i'll use pastebit |
| 06:23.55 | masked | bin* |
| 06:23.59 | parag_ast | kk |
| 06:24.00 | parag_ast | god |
| 06:24.02 | parag_ast | good |
| 06:25.09 | parag_ast | russellb, howz ur asterisknow project going on |
| 06:25.19 | parag_ast | i downloaded and found somany bugs... |
| 06:25.21 | parag_ast | sorry |
| 06:25.29 | russellb | heh, it's not my project |
| 06:25.35 | parag_ast | then |
| 06:25.36 | parag_ast | ?? |
| 06:25.38 | russellb | it's coming along, though |
| 06:25.46 | russellb | someone else at Digium is working on it ... |
| 06:25.57 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 06:26.23 | masked | http://pastebin.ca/273906 |
| 06:26.49 | masked | thats how it comes with the samples |
| 06:27.23 | SomeOne1 | Dec 10 02:27:29 NOTICE[15860]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 06:27.30 | SomeOne1 | god damnit this is pissing me off |
| 06:27.40 | masked | only thing i cant explain is why asterisk still tries to use that context if it doesn't exist or is not referenced |
| 06:27.53 | masked | SomeOne1: no route to destination? |
| 06:27.58 | masked | is the client running? |
| 06:28.02 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 06:28.21 | masked | try turning firewall off for testing? |
| 06:28.37 | masked | SomeOne1: what are you actually trying to dial? |
| 06:28.45 | masked | are they internal or external? |
| 06:29.05 | SomeOne1 | trying to dial out to a SIP server |
| 06:29.23 | masked | have you checked to see you are peered? |
| 06:29.41 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 06:29.48 | parag_ast | from wheree ${ARG2} |
| 06:29.50 | parag_ast | is comming |
| 06:29.51 | parag_ast | ?? |
| 06:29.53 | masked | ie sip show peers |
| 06:29.58 | masked | parag_ast: I have no idea. |
| 06:30.03 | masked | i can't find it anywhere |
| 06:30.12 | masked | i grepped /etc/asterisk and the source tree |
| 06:30.19 | parag_ast | no it must be somewhere in ur AGI |
| 06:30.21 | parag_ast | it seems |
| 06:30.28 | masked | so i guess asterisk must make it from the number dialed |
| 06:30.47 | parag_ast | not exectly |
| 06:30.50 | parag_ast | do one thing |
| 06:30.56 | parag_ast | u can change manually also |
| 06:31.02 | parag_ast | {arg2} |
| 06:32.34 | masked | pardon? |
| 06:32.38 | masked | do what one thing? |
| 06:32.46 | masked | i would have thought i could change it |
| 06:32.53 | parag_ast | something like exten = s,1,Dial(IAX2/100,30,t) |
| 06:32.54 | masked | i just haven't been able to find it |
| 06:33.26 | parag_ast | try it |
| 06:33.55 | parag_ast | and arg2 value is comming from AGI |
| 06:34.19 | masked | ok |
| 06:34.22 | masked | doing it manually works |
| 06:34.49 | parag_ast | good |
| 06:34.53 | parag_ast | now find out something |
| 06:34.54 | parag_ast | in |
| 06:35.02 | parag_ast | /var/lib/asterisk/agi/ |
| 06:35.15 | parag_ast | there will be script |
| 06:35.30 | parag_ast | which is deciding arg values |
| 06:36.27 | *** join/#asterisk SECG0D (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 06:36.58 | SomeOne1 | masked: i love you! |
| 06:36.59 | SomeOne1 | :P |
| 06:37.04 | SomeOne1 | it was a nat=no problem |
| 06:37.06 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 06:37.06 | masked | PPPoE0 agi-bin # grep -rn "ARG2" /var/lib/asterisk/agi-bin/ |
| 06:37.06 | masked | Binary file /var/lib/asterisk/agi-bin/eagi-test matches |
| 06:37.06 | masked | /var/lib/asterisk/agi-bin/recordingcheck:31:$type = $agi->get_variable("ARG2"); |
| 06:37.08 | masked | /var/lib/asterisk/agi-bin/dialparties.agi.pl:76:$dialopts = $AGI->get_variable('ARG2') || ''; |
| 06:37.08 | SomeOne1 | i changed it and it worked |
| 06:37.11 | masked | /var/lib/asterisk/agi-bin/dialparties.agi:52:$dialopts = get_var( $AGI, "ARG2" ); |
| 06:37.14 | masked | PPPoE0 agi-bin # |
| 06:37.27 | masked | SomeOne1: yeah you need nat on for sip |
| 06:38.45 | doolph | there's anything better than callprogress=yes ?? |
| 06:39.33 | masked | parag_ast: i don't see anything useful there |
| 06:40.43 | parag_ast | okk |
| 06:40.57 | parag_ast | i think dialparties.agi.pl |
| 06:42.34 | masked | $dialopts = $AGI->get_variable('ARG2') || ''; |
| 06:42.34 | masked | ? |
| 06:42.44 | masked | thats unfamiliar to me |
| 06:42.46 | masked | i dont know perl |
| 06:42.53 | parag_ast | hehehe |
| 06:42.59 | parag_ast | even i don't |
| 06:43.01 | parag_ast | :) |
| 06:43.10 | masked | it doesn't look useful anyway |
| 06:43.13 | parag_ast | okk do one thing |
| 06:43.57 | parag_ast | can u go into /etc/asterisk/iax.conf |
| 06:44.03 | parag_ast | and find 1337 user |
| 06:44.08 | parag_ast | and find out the context |
| 06:44.14 | parag_ast | and tell me the context |
| 06:45.00 | masked | there was a typo, i'd used a k instead of an x in context |
| 06:45.34 | masked | ok |
| 06:45.38 | masked | so that works |
| 06:46.02 | parag_ast | good |
| 06:46.03 | parag_ast | hehehe |
| 06:46.11 | masked | if i set to default context, but why was it calling on the marco-stdexten when one wasn't set? |
| 06:46.48 | parag_ast | yeh may be if non context selected then it goes to macro-stdexten |
| 06:47.03 | parag_ast | so because of ur typo mistake |
| 06:47.08 | parag_ast | it was not able to identify |
| 06:47.12 | masked | :S |
| 06:47.14 | masked | ahh well |
| 06:47.18 | parag_ast | and it was going to macro-stdexten |
| 06:47.20 | masked | thats good enough for now |
| 06:47.41 | masked | yeah i just thought it would have tried to dial say 1337@ |
| 06:47.48 | masked | and say no context |
| 06:47.51 | parag_ast | heheh |
| 06:47.53 | parag_ast | right |
| 06:47.54 | masked | but it actually was going somewhere |
| 06:48.07 | parag_ast | hmm DIALPLANS ARE HEART OF ASTERISK |
| 06:48.16 | masked | heh |
| 06:53.40 | *** part/#asterisk Mario (n=Mario@69-161-97-50.bflony.adelphia.net) |
| 06:55.29 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 06:56.06 | *** part/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 06:58.07 | masked | weird |
| 06:58.41 | masked | as an iax client, it doesn't read context= from users.conf but only iax.conf |
| 06:58.59 | masked | where as sip clients read from users.cong |
| 06:59.00 | parag_ast | Yup |
| 06:59.07 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 07:00.58 | masked | that makes things difficult |
| 07:01.50 | masked | all the same |
| 07:01.56 | masked | the marco still doesn't work |
| 07:02.04 | masked | have to use hard coded extensions |
| 07:11.20 | *** join/#asterisk jerlique2 (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 07:11.45 | *** part/#asterisk jerlique2 (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 07:12.07 | *** join/#asterisk arley (n=arley@66-214-136-53.dhcp.gldl.ca.charter.com) |
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| 07:16.23 | *** join/#asterisk corleno (n=corloene@60.53.36.33) |
| 07:16.47 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:21.31 | *** join/#asterisk arley (n=arley@66-214-136-53.dhcp.gldl.ca.charter.com) |
| 07:22.13 | SplasPood | is there any way to define a timeout for CURL() ? |
| 07:26.01 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 07:28.09 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-50-134.w90-5.abo.wanadoo.fr) |
| 07:30.28 | *** join/#asterisk Un1x_Laptop (i=Sean@CPE0051c5534c1f-CM001371145406.cpe.net.cable.rogers.com) |
| 07:32.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:35.52 | *** join/#asterisk ToTo (n=ToTo@host90-161-dynamic.0-87-r.retail.telecomitalia.it) |
| 07:39.11 | parag_ast | anybody know's any good perl editor |
| 07:41.09 | doolph | anyone can guide me about callprogress/answer detection problems? |
| 07:44.07 | *** join/#asterisk ComPuTeR (n=eNd@88.224.164.25) |
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| 08:29.21 | shellshark | re |
| 08:29.28 | shellshark | parag_ast: vim ;) |
| 08:48.36 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 08:50.46 | robin__sz | parag_ast, you need a good Perl editor? |
| 08:51.06 | robin__sz | parag_ast, vim is pretty much ideal |
| 08:52.05 | robin__sz | parag_ast, or emacs of you prefer ... gedit has Perl syntax highlighting too, if you prefer a more GUI sort fo approach. |
| 08:54.25 | zapp-branigan | hi , i have this warning chan_zap.c:10874 setup_zap: Ignoring switchtype |
| 08:54.37 | zapp-branigan | what is the problem ? |
| 08:54.58 | hads | Normal |
| 08:54.59 | zapp-branigan | and the zaps do not work |
| 08:55.09 | zapp-branigan | :( |
| 08:55.34 | zapp-branigan | hands is normal ? |
| 08:59.16 | *** join/#asterisk LakeSolon (n=blake@64-83-212-33.dhcp.stcd.mn.charter.com) |
| 09:05.44 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
| 09:07.13 | [hC] | So ive got presence turned on on my polycom 501's for hint support, and two softkeys are there, 'mystat' and 'buddies' - makes sense. is there a way to leave presence on but turn these soft keys off? they dont have anything to do w./ asterisk and just confuse people |
| 09:09.12 | [hC] | maybe a better question - CAN this stuff interop w/ asterisk? and does it need to be on phones that say, a receptionist monitors (the 501s for example, since they have no BLF) |
| 09:09.13 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 09:09.19 | [hC] | would they need to have presence on to be MONITORED? |
| 09:09.57 | *** part/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 09:13.31 | *** join/#asterisk brush01uk (n=brush01u@62-30-75-207.cable.ubr03.brom.blueyonder.co.uk) |
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| 09:21.03 | zapp-branigan | hi what is the way to text the zap chanels ? |
| 09:21.16 | zapp-branigan | the internals calls work |
| 09:21.30 | zapp-branigan | but the zap call don't work |
| 09:21.53 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 09:22.13 | zapp-branigan | i have istalled the zaptel and ztcfg work fine |
| 09:22.21 | zapp-branigan | :? |
| 09:24.29 | zapp-branigan | Dec 10 10:23:32 NOTICE[9925]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 09:24.29 | zapp-branigan | <PROTECTED> |
| 09:28.14 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 09:29.24 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 09:30.39 | parag_ast | zapp-branigan, which card r u using |
| 09:30.45 | parag_ast | is it TDM400P |
| 09:33.02 | shellshark | zapp-branigan: very, eh, "original" nickname ;) |
| 09:33.21 | parag_ast | HEHEHE |
| 09:33.56 | parag_ast | okk now a stupid question from my side....is there any way to listen online conversation |
| 09:34.14 | parag_ast | i mean the people who are talking threw asterisk box |
| 09:34.23 | parag_ast | can we listen like MAN IN MIDDLE |
| 09:34.40 | parag_ast | LIVE CHANNELS |
| 09:35.41 | dlynes_laptop | parag_ast: dood...long time, no see! |
| 09:35.51 | dlynes_laptop | parag_ast: it's called chanspy |
| 09:36.02 | parag_ast | hey dlynes.... |
| 09:36.07 | parag_ast | how r u dear |
| 09:36.20 | dlynes_laptop | good |
| 09:36.25 | dlynes_laptop | Just busy with work and such |
| 09:36.30 | parag_ast | yeh |
| 09:36.34 | dlynes_laptop | Working on a new module for asterisk, too |
| 09:36.40 | dlynes_laptop | So that you can ring multiple extensions easily |
| 09:36.59 | dlynes_laptop | Currently, if you want to ring multiple extensions, you have to use the dial command |
| 09:37.11 | dlynes_laptop | And that's not terribly flexible for inuse handling |
| 09:37.12 | parag_ast | yeh |
| 09:37.29 | parag_ast | but then also u can make ring groups |
| 09:37.30 | parag_ast | right |
| 09:37.32 | dlynes_laptop | I didn't want to have to write a 300 line dialplan code |
| 09:37.40 | parag_ast | so u are writing application exectly for ring groups |
| 09:37.44 | dlynes_laptop | So I decided to write an application module, instead |
| 09:37.48 | dlynes_laptop | Nah |
| 09:37.54 | dlynes_laptop | ring groups are something different |
| 09:38.07 | dlynes_laptop | asterisk already supports ring groups |
| 09:38.12 | parag_ast | yup |
| 09:38.17 | parag_ast | thats really good |
| 09:38.32 | dlynes_laptop | Just by passing multiple peers along the dial command |
| 09:38.36 | dlynes_laptop | This is pretty much like that |
| 09:38.42 | parag_ast | ohh okk |
| 09:38.44 | dlynes_laptop | But it allows you greater flexibility on the ring groups |
| 09:38.53 | dlynes_laptop | say for example: |
| 09:40.27 | dlynes_laptop | exten => s,1,GetAllAvail(SIP/101\(_[2-5]\)*&SIP/102\(_[2-5]\)*&SIP/103\(_[2-5]\)*&SIP/104\(_[2-5]\)*&SIP/105\(_[2-5]\)*) |
| 09:41.00 | dlynes_laptop | exten => s,2,Dial(${CHANSAVAIL}) |
| 09:41.10 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 09:41.25 | parag_ast | so it will ring all |
| 09:41.26 | parag_ast | right |
| 09:41.37 | parag_ast | if they are avaliable |
| 09:41.45 | dlynes_laptop | So it'll use the regex for each member of the ring group to define what peers should be considered to be counted as the member |
| 09:42.03 | dlynes_laptop | and it'll grab the first available peer for each member |
| 09:42.08 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 09:42.37 | parag_ast | excellent job man |
| 09:43.31 | dlynes_laptop | I take it you'd be interested in such a module, too? |
| 09:44.00 | parag_ast | Yeh sureee |
| 09:44.30 | hads | dlynes_laptop: Are you releasing the app? |
| 09:44.40 | parag_ast | nopp |
| 09:44.43 | dlynes_laptop | hads: I will be, yes |
| 09:44.58 | dlynes_laptop | hads: I'll be making it work on both asterisk and openpbx |
| 09:45.05 | hads | Nice. |
| 09:45.16 | dlynes_laptop | hads: asterisk will be first, because that's what all my customers are currently using |
| 09:45.22 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 09:45.31 | hads | Understandable. |
| 09:46.45 | parag_ast | dlynes_laptop, pls give me some idea of chan_spy |
| 09:46.48 | zapp-branigan | hi parag_ast my card is digium 4 chanels pci |
| 09:46.50 | parag_ast | i never used it |
| 09:46.53 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 09:47.10 | dlynes_laptop | parag_ast: neither have I |
| 09:47.16 | dlynes_laptop | parag_ast: you can always check the wiki, though |
| 09:47.18 | parag_ast | is it successful |
| 09:47.23 | parag_ast | ?? |
| 09:47.30 | dlynes_laptop | parag_ast: well, lots of people are using it in call center applications |
| 09:47.34 | parag_ast | i need to monitor IAX2 channels |
| 09:47.52 | dlynes_laptop | parag_ast: there's two spy applications, too |
| 09:48.04 | dlynes_laptop | parag_ast: one that's a channel-specific spy, and one that's generic |
| 09:48.23 | parag_ast | ohh okk |
| 09:50.12 | zapp-branigan | dlynes_laptop getallavails comand can not be found in google |
| 09:50.25 | zapp-branigan | where you have see this ? |
| 09:50.32 | dlynes_laptop | zapp-branigan: i'm writing it |
| 09:50.37 | dlynes_laptop | zapp-branigan: it doesn't exist yet |
| 09:50.43 | zapp-branigan | :P |
| 09:51.00 | zapp-branigan | is a good function |
| 09:51.21 | zapp-branigan | and how you detect this ? |
| 09:51.43 | zapp-branigan | by status line ? |
| 09:51.43 | dlynes_laptop | I've written some C++ code to handle that |
| 09:51.57 | dlynes_laptop | It'll be called app_getallavail.so |
| 09:52.13 | zapp-branigan | its good |
| 09:52.28 | dlynes_laptop | So there'll be some C++ code and a Makefile |
| 09:52.33 | dlynes_laptop | So you just type 'make' |
| 09:52.36 | dlynes_laptop | and then 'make install' |
| 09:52.51 | dlynes_laptop | Then in the asterisk cli, type 'load app_getallavail.so' |
| 09:53.46 | zapp-branigan | work or not ? |
| 09:54.15 | dlynes_laptop | Not yet |
| 09:54.26 | zapp-branigan | :( |
| 09:54.26 | dlynes_laptop | Like I said...I'm still working on it |
| 09:54.26 | zapp-branigan | what is the problem |
| 09:54.56 | dlynes_laptop | I've got it to the point where it's walking through all the locked channels and parsing the application parameters |
| 09:54.59 | dlynes_laptop | No problem |
| 09:55.05 | dlynes_laptop | I just haven't finished the code yet |
| 09:55.22 | zapp-branigan | :D |
| 09:58.11 | zapp-branigan | and where you will place the module ? some web ? |
| 09:59.13 | dlynes_laptop | Yeah...I'll probably make it downloadable from my company's webserver, initially |
| 09:59.39 | shellshark | idlerpg pwns |
| 09:59.42 | zapp-branigan | :O |
| 10:00.53 | zapp-branigan | when you have some beta told me to text :P |
| 10:01.49 | dlynes_laptop | zapp-branigan: yeah the first version I put up won't have regex support, probably |
| 10:02.02 | dlynes_laptop | zapp-branigan: or if it does, it'll be limited, not full support |
| 10:02.28 | zapp-branigan | ok |
| 10:02.55 | shellshark | dlynes_laptop: what are you writing? |
| 10:03.32 | shellshark | getallavails does what exactly? |
| 10:03.50 | zapp-branigan | if i can make work the zaptel i will text your module :P |
| 10:04.02 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 10:04.19 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 10:04.51 | dlynes_laptop | shellshark: It'll allow you to define flexible ring groups, using regex syntax |
| 10:05.06 | dlynes_laptop | shellshark: however, after I get the regex in there |
| 10:05.19 | shellshark | "flexible ring groups" ? |
| 10:05.30 | dlynes_laptop | shellshark: I might not be able to contribute it back to asterisk...I might have to make it separately available, because of licensing on the regex library |
| 10:05.53 | dlynes_laptop | shellshark: Scroll back up to see my example |
| 10:06.02 | shellshark | pcre has a gpl license? |
| 10:06.10 | dlynes_laptop | no idea |
| 10:06.29 | dlynes_laptop | but the thing is, I don't know if whatever library I use for regex parsing will be compatible with asterisk's licensing |
| 10:06.45 | shellshark | pcre would be |
| 10:06.46 | dlynes_laptop | I don't want to reinvent the wheel when it comes to regex parsing |
| 10:06.55 | shellshark | pcre man, pcre ;) |
| 10:10.45 | *** join/#asterisk ComPuTeR (n=BLaCkGir@88.224.164.25) |
| 10:15.07 | dlynes_laptop | Yeah, another reason i don't think digium will accept the code anyways, is because i'm writing it in C++ :) |
| 10:15.15 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 10:16.38 | zapp-branigan | why ? |
| 10:16.55 | zapp-branigan | he dot program in c++? |
| 10:17.42 | zapp-branigan | dlynes_laptop digium only use c not c++ ? |
| 10:17.48 | zapp-branigan | or some else ?¿ |
| 10:17.59 | dlynes_laptop | que? |
| 10:18.06 | dlynes_laptop | he dot program? |
| 10:18.07 | dlynes_laptop | huh? |
| 10:18.10 | zapp-branigan | como que que |
| 10:18.36 | dlynes_laptop | Yes, digium only uses C |
| 10:18.39 | zapp-branigan | dlynes_laptop you speak spanish? |
| 10:18.50 | dlynes_laptop | I just like using C++ because it makes the code more readable and more maintainable |
| 10:18.59 | dlynes_laptop | No, senor :) |
| 10:19.05 | zapp-branigan | for the functions |
| 10:19.34 | zapp-branigan | you call a class |
| 10:19.38 | dlynes_laptop | I speak English fluently, and I can almost communicate on a very basic level in Mandarin |
| 10:19.48 | zapp-branigan | hooo |
| 10:19.49 | zapp-branigan | :P |
| 10:20.07 | dlynes_laptop | and I used to somewhat fluent in French |
| 10:20.18 | dlynes_laptop | but I haven't used French in so many years that I lost most of it |
| 10:20.24 | zapp-branigan | que is espanish to ask what hapend |
| 10:20.40 | zapp-branigan | <dlynes_laptop> que? |
| 10:21.14 | dlynes_laptop | Oh...I thought 'que' meant 'what' :) |
| 10:21.22 | zapp-branigan | yes |
| 10:21.27 | zapp-branigan | the same |
| 10:21.45 | parag_ast | Its Hindi word |
| 10:21.47 | parag_ast | isn't it |
| 10:21.52 | dlynes_laptop | maybe |
| 10:21.54 | zapp-branigan | i too write the programs in c++ |
| 10:21.58 | dlynes_laptop | but it's also Spanish and French, too |
| 10:22.07 | zapp-branigan | but i use the visual c++ .net |
| 10:22.07 | parag_ast | but meaning is same |
| 10:22.20 | dlynes_laptop | Yeah..I write C# and Java code, too |
| 10:22.43 | zapp-branigan | the c# do not like nothing |
| 10:22.45 | dlynes_laptop | and perl, and C, and C++, and bash shell scripting, and VB, and ... |
| 10:22.53 | dlynes_laptop | I just write code in general |
| 10:22.59 | dlynes_laptop | The language really doesn't mean much to me |
| 10:23.00 | zapp-branigan | perl too and php |
| 10:23.04 | zapp-branigan | vhdl |
| 10:23.21 | zapp-branigan | and microcontrollers all |
| 10:23.21 | dlynes_laptop | Unfortunately though, I haven't had an opportunity to write in assembly language for a while |
| 10:23.28 | dlynes_laptop | I really miss writing code in assembly |
| 10:23.54 | dlynes_laptop | Last project I had the opportunity to use it in was a full screen real-time video conferencing engine |
| 10:23.59 | zapp-branigan | i write all the assenble code of my live in the university |
| 10:24.05 | zapp-branigan | and never more |
| 10:24.06 | zapp-branigan | :P |
| 10:24.26 | zapp-branigan | i only use now c |
| 10:24.31 | dlynes_laptop | Previous project was a real time stock market reporting engine; the entire server was written in 80386 protected mode assembly language |
| 10:24.40 | dlynes_laptop | it wasn't poisoned by any high level languages at all |
| 10:25.08 | dlynes_laptop | It was about 100K lines of 80386 protected mode assembly language code |
| 10:25.11 | zapp-branigan | i use the ensambler of intel |
| 10:25.26 | dlynes_laptop | We wrote it in a mix of Borland Turbo Assembler and Microsoft Macro Assembler |
| 10:25.43 | zapp-branigan | and do not like only jump 1024 bytes |
| 10:25.50 | dlynes_laptop | Microsoft's assembler is pretty nice |
| 10:26.03 | dlynes_laptop | It's one of hte few products they put out that I actually liked |
| 10:26.11 | zapp-branigan | when i write a jump i must do jumps along of the code |
| 10:27.05 | zapp-branigan | to go from the start to the end of the code in jump i must to insert little jump along the code |
| 10:27.39 | zapp-branigan | 8086 must be in the trash |
| 10:28.18 | zapp-branigan | and make a new controller from 0 |
| 10:28.35 | zapp-branigan | humm |
| 10:29.06 | *** join/#asterisk budmang (i=budman@12.206.134.162) |
| 10:29.09 | budmang | anyonein? |
| 10:29.18 | zapp-branigan | zapp-branigan make miny jump to avoid dlynes_laptop fall |
| 10:29.28 | dlynes_laptop | heh |
| 10:29.32 | zapp-branigan | hum |
| 10:29.35 | dlynes_laptop | budmang: nope...just us bots here |
| 10:29.50 | zapp-branigan | i'm sorry for my english :) |
| 10:30.11 | dlynes_laptop | zapp-branigan: s/miny/mini/ |
| 10:30.31 | dlynes_laptop | zapp-branigan: you meant 'small', right? |
| 10:30.43 | zapp-branigan | yes |
| 10:30.46 | shellshark | zapp-branigan: you should appologize for stealing your nick ;) |
| 10:30.58 | budmang | I need a good provider. |
| 10:30.59 | budmang | :-) |
| 10:31.03 | zapp-branigan | hi i want to text the zap chanels who can do this? |
| 10:31.07 | shellshark | budmang: shellshark.net :) |
| 10:31.10 | dlynes_laptop | budmang: try www.calltermination.com |
| 10:31.11 | zapp-branigan | by php by a module ? |
| 10:31.54 | zapp-branigan | i refear the dial plan by a c code |
| 10:31.58 | budmang | i currently use teliax pay by minute but i want a good unlimited plan or 500 mins. |
| 10:31.59 | zapp-branigan | can be do this ? |
| 10:32.30 | shellshark | budmang: we have unlimited calling starting at $15/mo |
| 10:32.36 | shellshark | budmang: to the US and Canada |
| 10:33.11 | zapp-branigan | dlynes_laptop where you have see to make the modules |
| 10:33.15 | budmang | shellshark |
| 10:33.17 | budmang | pm me pleae |
| 10:33.19 | budmang | please |
| 10:33.23 | *** join/#asterisk lib (n=private@cpe-76-168-249-132.socal.res.rr.com) |
| 10:33.24 | zapp-branigan | i want make modules too |
| 10:33.46 | dlynes_laptop | budmang: you can also try http://www.voip-info.org/wiki/ and do a search for 'voip service providers' |
| 10:34.12 | dlynes_laptop | zapp-branigan: how to make the modules? |
| 10:34.30 | dlynes_laptop | zapp-branigan: check the source code for you version of asterisk, and then look in the doc directory |
| 10:34.37 | zapp-branigan | the same how you are doing |
| 10:34.41 | dlynes_laptop | zapp-branigan: there's a few files in there...you can also do make progdocs |
| 10:34.56 | zapp-branigan | ok |
| 10:35.05 | dlynes_laptop | zapp-branigan: and it'll build the doxygen documentation in /path/to/asterisk/sources/doc/api/html/index.html |
| 10:35.22 | zapp-branigan | :O |
| 10:35.26 | dlynes_laptop | zapp-branigan: the ones in the doc directory tell you the coding guidelines |
| 10:35.37 | dlynes_laptop | zapp-branigan: the doxygen documentation gives you full documentation on all the code |
| 10:35.44 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 10:35.51 | dlynes_laptop | zapp-branigan: I think there might also be a sample skeletal app in there somewhere, too |
| 10:37.42 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 10:39.19 | zapp-branigan | yes i have found that |
| 10:40.33 | zapp-branigan | hi another question my asterisk can receive user in the same net but can not receive from internet |
| 10:40.43 | zapp-branigan | and i have open the ports |
| 10:40.47 | zapp-branigan | in the router |
| 10:40.57 | zapp-branigan | iptables -A INPUT -p tcp --dport 80 -j ACCEPT |
| 10:40.57 | zapp-branigan | iptables -A OUTPUT -p tcp --sport 80 -j ACCEPT |
| 10:40.57 | zapp-branigan | iptables -A OUTPUT -p udp --sport 5060 -j ACCEPT |
| 10:40.57 | zapp-branigan | iptables -A INPUT -p udp --sport 5060 -j ACCEPT |
| 10:40.57 | zapp-branigan | iptables -A INPUT -p udp --dport 10000:20000 -j ACCEPT |
| 10:40.57 | zapp-branigan | iptables -A OUTPUT -p udp --dport 10000:20000 -j ACCEPT |
| 10:40.59 | zapp-branigan | iptables -A INPUT -p udp --sport 4569 -j ACCEPT |
| 10:41.01 | zapp-branigan | iptables -A OUTPUT -p udp --sport 4569 -j ACCEPT |
| 10:41.03 | zapp-branigan | and linux |
| 10:41.07 | zapp-branigan | Router and linux |
| 10:41.07 | shellshark | use a pastbin! |
| 10:41.12 | shellshark | pastebin.ca |
| 10:41.25 | zapp-branigan | what is this ? |
| 10:41.51 | zapp-branigan | haaa |
| 10:42.30 | zapp-branigan | i must to add some another port ? |
| 10:44.51 | *** join/#asterisk ComPuTeR (n=_figen_@88.224.164.25) |
| 10:46.17 | *** join/#asterisk CleanerX (n=nix@p54A3A447.dip0.t-ipconnect.de) |
| 10:47.30 | JT | zapp-branigan: it's rude to paste lots of lines to the channel |
| 10:48.06 | dlynes_laptop | ~pb |
| 10:48.12 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:48.53 | dlynes_laptop | zapp-branigan: I find it's easier once you start getting complicated setups for the firewall to use a configuration tool that allows flexibility, but is still easy to use |
| 10:49.04 | dlynes_laptop | zapp-branigan: I would suggest using something like shorewall to configure your firewall |
| 10:49.12 | dlynes_laptop | zapp-branigan: you'll save yourself a lot of grief |
| 10:49.44 | dlynes_laptop | zapp-branigan: www.shorewall.net |
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| 11:07.48 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 11:35.50 | Marshall16 | what's a good VOIP IAX2 service provider? |
| 11:36.48 | dlynes_laptop | Marshall16: most popular seems to be teliax |
| 11:36.59 | dlynes_laptop | Marshall16: but there's several iax2 providers |
| 11:37.29 | Marshall16 | can you list a few? |
| 11:37.36 | dlynes_laptop | Not offhand, no |
| 11:37.40 | dlynes_laptop | But then again, I don't use them |
| 11:37.54 | dlynes_laptop | JerJer works for one, too |
| 11:38.02 | dlynes_laptop | But I can't remember what the name of it is offhand |
| 11:38.18 | dlynes_laptop | And there's another guy on here that owns one |
| 11:40.29 | dlynes_laptop | The one that JerJer works for is the one that's been around the longest too |
| 11:48.35 | zoa | jerjer = nufone |
| 11:48.48 | zoa | voiptalk.org should also be fine |
| 11:48.50 | zoa | and voxbone |
| 11:49.08 | zoa | voipgate is also quite big |
| 11:49.14 | Marshall16 | anyone got an iax2 account i can use? |
| 11:49.14 | Marshall16 | ;\ |
| 11:51.51 | dlynes_laptop | Oh yeah...nufone that's what it was |
| 11:52.37 | dlynes_laptop | and the guy that's known as the biggest promoter of voip used to have an iax2 server...I don't know if he still has, or not |
| 11:53.09 | dlynes_laptop | canm't remember his name or his company's name offhand, either :( |
| 11:54.17 | zoa | fwd ? |
| 11:58.08 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-f8dfe7707c84e221) |
| 11:58.59 | *** join/#asterisk mega (n=mega@217.201.192.15) |
| 12:03.01 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 12:05.20 | *** join/#asterisk ltd (n=z@202-161-13-39.dyn.iinet.net.au) |
| 12:08.16 | dlynes_laptop | Oh yeah..pulver.com |
| 12:08.36 | parag_ast | pulver.com is very famous man for IAX termination |
| 12:08.44 | parag_ast | they only launched fwdout |
| 12:08.46 | parag_ast | FWDOUT |
| 12:12.53 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
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| 12:18.13 | *** join/#asterisk Omer (i=Omer@202.133.79.19) |
| 12:19.17 | *** join/#asterisk qwertz (n=qwertz@67.206.27.217.static.versanetonline.de) |
| 12:21.10 | qwertz | Hi, playing with the queues function atm, if i call the queue number all agents ring, but after some time I get "Got SIP response 486 "Busy Here"" in the cli and the agent phones stop ringing - does anybody know what I' |
| 12:21.28 | qwertz | Hi, playing with the queues function atm, if i call the queue number all agents ring, but after some time I get "Got SIP response 486 "Busy Here"" in the cli and the agent phones stop ringing - does anybody know what I'm doing wrong? |
| 12:22.17 | Omer | can i have the Chanspy code? |
| 12:22.31 | Omer | for spying selected sip channels |
| 12:29.31 | eliXier | ahh damn, my pc freeze... |
| 12:29.51 | eliXier | my question: has the wl-500gD under RC6 a failsafe-mode? |
| 12:30.27 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 12:33.13 | dlynes_laptop | Omer: chanspy code? chanspy should already be included with your asterisk code |
| 12:34.11 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 12:43.44 | Omer | yes |
| 12:44.09 | Omer | but i remmebr i need to add something more |
| 12:44.31 | Omer | i reinstalled my box and lost the code :S |
| 12:46.42 | dlynes_laptop | Omer: ftp://ftp.digium.com/pub/telephony/asterisk/asterisk-1.2.13.tar.gz |
| 12:50.04 | Omer | thats an asterisk ? |
| 12:50.36 | dlynes_laptop | yes |
| 12:50.41 | dlynes_laptop | that's _THE_ asterisk |
| 12:50.49 | dlynes_laptop | It's the source code for asterisk |
| 12:51.08 | dlynes_laptop | Which ideally is what you really should be using, anyways |
| 12:51.22 | dlynes_laptop | Then you don't have to worry about whether your linux distribution has packaged asterisk properly or not |
| 12:52.30 | Omer | oh ok |
| 12:52.56 | Omer | my office people told me that they want to spy on sip channels |
| 12:53.09 | Omer | and i m juststuck how do i gave them that option |
| 12:53.18 | Omer | to spy on selected channels |
| 12:53.22 | dlynes_laptop | ~wiki |
| 12:53.24 | Omer | i use to dial 888 |
| 12:53.28 | Omer | and it works |
| 12:53.30 | Omer | ok |
| 12:53.32 | dlynes_laptop | ~wikis |
| 12:53.34 | jbot | from memory, wikis is http://www.voip-info.org |
| 12:53.44 | dlynes_laptop | There, look for asterisk |
| 12:53.47 | dlynes_laptop | Then applications |
| 12:53.53 | Omer | ok cool |
| 12:53.53 | dlynes_laptop | Then chanspy |
| 12:53.54 | Omer | thans |
| 12:53.56 | Omer | thanks |
| 12:54.05 | Omer | btw i need to asterisk consultants |
| 12:54.15 | Omer | for remote support |
| 12:54.17 | dlynes_laptop | You can find those on the wiki, too |
| 12:54.23 | Omer | oh cool |
| 12:54.38 | Omer | i removed mitel with asterisk |
| 12:54.39 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 12:54.48 | Omer | as asterisk gives more options :P |
| 12:55.51 | Omer | is it fine to put asterisk box ob public ip |
| 12:55.53 | Omer | ? |
| 12:55.59 | Omer | or will i get hacked every time? |
| 12:56.14 | dlynes_laptop | I've got a number of asterisk boxes on public ips |
| 12:56.22 | dlynes_laptop | But I also made sure I secured them, too |
| 12:56.24 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 12:56.39 | Omer | then you must be into programing too much |
| 12:57.09 | Omer | i have worked on MS platforms then directly switch to asterisk |
| 12:57.10 | Omer | :P |
| 12:57.29 | dlynes_laptop | I work on Windows, Linux and Solaris |
| 12:57.36 | Omer | cool |
| 12:57.37 | dlynes_laptop | I used to work on OS/2 as well |
| 12:57.57 | dlynes_laptop | Not to mention Windows 3.x, MSDOS, DRDOS, and PCDOS |
| 12:58.04 | Omer | i have 50 user with recording and i am using 3.0 Ghz intel with 4 GB Ram |
| 12:58.06 | Omer | will it work fine |
| 12:58.07 | Omer | ?? |
| 12:58.21 | Omer | dem you have worked on too many things |
| 12:58.43 | dlynes_laptop | It might work fine and it might not |
| 12:58.48 | dlynes_laptop | I don't really know offhand |
| 12:59.02 | Omer | ook |
| 12:59.18 | dlynes_laptop | There's a number of items on the wiki that give you examples for some existing configurations that work well for certain requirements |
| 12:59.28 | Omer | yes |
| 12:59.40 | dlynes_laptop | It's also going to depend on which 3.0Ghz intel processor, too |
| 12:59.41 | Omer | do you think replacing mitel with asterisk will be a good idea |
| 12:59.43 | Omer | ? |
| 12:59.49 | Omer | oh ok |
| 12:59.50 | dlynes_laptop | Intel has about ten different processors that are 3.0Ghz |
| 13:00.08 | dlynes_laptop | replacing mitel with asterisk is always a good idea |
| 13:00.16 | Omer | goood :P |
| 13:00.27 | dlynes_laptop | The person doing the replacing might always be a good idea though :) |
| 13:00.31 | dlynes_laptop | erm |
| 13:00.36 | dlynes_laptop | might not :) |
| 13:00.53 | Omer | well i dont like mitel thats why im replacing with mitel |
| 13:01.04 | Omer | and my boss gave me some tasks to complete |
| 13:01.18 | dlynes_laptop | Well, me |
| 13:01.24 | Omer | i have done evry thing so ofar just got stuck on spying on selected channels |
| 13:01.25 | Omer | :s |
| 13:01.25 | dlynes_laptop | I just don't like mitel vendors |
| 13:01.30 | dlynes_laptop | that's why i want to replace them |
| 13:01.34 | Omer | helyea neither do i |
| 13:01.47 | dlynes_laptop | they have a lock on the hotel industry |
| 13:01.54 | Omer | ys |
| 13:01.56 | dlynes_laptop | it's time to break that lock |
| 13:02.12 | Omer | they charged me 300$ for recording for per channel basis |
| 13:02.17 | Omer | yes |
| 13:02.32 | Omer | im promoting asterisk in pk |
| 13:02.52 | Omer | got 10 call centers to switch on it already |
| 13:03.52 | dlynes_laptop | Yeah....lots of pakistanis here |
| 13:03.57 | Omer | really |
| 13:04.02 | dlynes_laptop | Lots of Indians, too |
| 13:04.08 | Omer | sounds good |
| 13:05.17 | Omer | where you from |
| 13:05.17 | Omer | ? |
| 13:05.30 | dlynes_laptop | Vancouver, Canada |
| 13:05.46 | dlynes_laptop | anyways |
| 13:05.48 | dlynes_laptop | gotta run |
| 13:05.49 | dlynes_laptop | need sleep |
| 13:05.55 | dlynes_laptop | talk to you later |
| 13:06.08 | dlynes_laptop | Try talking to Dr-Linux|work sometime |
| 13:06.18 | dlynes_laptop | He's in Lahore |
| 13:06.32 | parag_ast | Indian **** |
| 13:06.35 | parag_ast | hehehe |
| 13:06.35 | dlynes_laptop | He works at a call center, too |
| 13:06.39 | dlynes_laptop | And parag_ast is Indian :) |
| 13:06.46 | parag_ast | Yehhh |
| 13:07.21 | dlynes_laptop | and Joel is in Gujarat |
| 13:08.15 | dlynes_laptop | and Yacko is in Mumbai I think |
| 13:08.44 | dlynes_laptop | and DaeJeon is in Korea, but he's from Punjab |
| 13:09.04 | parag_ast | good to know man |
| 13:09.12 | dlynes_laptop | DaeJeon doesn't hang out here |
| 13:09.16 | dlynes_laptop | He just hangs out in #solaris |
| 13:09.26 | dlynes_laptop | He's trying to get asterisk up and running on Solaris |
| 13:09.51 | dlynes_laptop | but i've been trying to convince him to come on here tonight |
| 13:10.16 | dlynes_laptop | anyways |
| 13:10.19 | dlynes_laptop | heading to sleep |
| 13:10.21 | dlynes_laptop | good night |
| 13:10.50 | parag_ast | good night |
| 13:10.53 | parag_ast | dlynes |
| 13:10.56 | parag_ast | and thanks for all info |
| 13:13.14 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:13.16 | Omer | is his name junid? |
| 13:14.43 | parag_ast | Junid |
| 13:14.44 | parag_ast | ?? |
| 13:15.31 | Omer | junaid upal |
| 13:16.45 | *** join/#asterisk ltd (n=z@202-161-13-39.dyn.iinet.net.au) |
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| 13:25.39 | *** join/#asterisk pvr (n=pvr@pluto.anders.ru) |
| 13:26.30 | pvr | hi ppl |
| 13:28.34 | pvr | had anyone such problem - phone, connected to addpac, is ringing one time, and don't ring anymore |
| 13:28.39 | pvr | but when hang up, call leg is connectiong normaly |
| 13:29.17 | pvr | call comes from oh323 via * to addpac(sip) |
| 13:32.04 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.189) |
| 13:33.43 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
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| 13:50.25 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 13:50.34 | qwertz | Hi, I've got * 1.0.10 running with a queue and agents. the agents have snom300 phones now I'd like agents to logon just by a keypress. atm it's possible for the agents to call 999 to logon - they don't have to do anything. is there a way to do this with the snom function keys? |
| 13:53.44 | *** join/#asterisk Explisit (n=explisit@85.187.163.57) |
| 13:55.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:55.54 | Explisit | Hello, I have problem with my Asterisk. Everything works just fine, but when I make a call from ip phone elesign-1202 there is no voice in either direction. When I call from other ip phones to elesign-1202 everything is in order. Can anyone guide me to some solution to this problem. Thanks :) |
| 13:58.02 | parag_ast | this is due to firewall |
| 13:58.12 | parag_ast | that means ur firewall is stopping RTP Ports |
| 13:58.26 | Explisit | there is no firewall |
| 13:58.34 | parag_ast | just try with NAT=yes and allow 10000-20000 |
| 13:59.07 | parag_ast | okk that means extention to extention calling is working |
| 13:59.07 | parag_ast | right |
| 13:59.15 | Explisit | well the phones are not behind nat and iptables -F |
| 13:59.23 | parag_ast | okk |
| 13:59.52 | Explisit | also when the other side pick up asterisk says something like - the connetion must re-setup |
| 14:00.54 | Explisit | this is the only thing that bother me. it's with Warning level |
| 14:01.56 | parag_ast | cli:> sip show registry |
| 14:02.04 | Explisit | oh323 |
| 14:02.05 | parag_ast | and check if sip phones are registered or not |
| 14:02.11 | parag_ast | ohh u are using oh323 |
| 14:02.12 | parag_ast | ?? |
| 14:02.16 | Explisit | yes |
| 14:02.34 | Explisit | with h323 or ooh323 the thing are very bad :) |
| 14:03.00 | parag_ast | yehhh |
| 14:03.05 | parag_ast | seriously speaking |
| 14:03.14 | parag_ast | H323 dosn't work well |
| 14:03.17 | parag_ast | with asterisk |
| 14:03.20 | parag_ast | believe me |
| 14:03.58 | Explisit | ok but i need it |
| 14:04.20 | parag_ast | sorry then no idea boss |
| 14:08.19 | *** join/#asterisk xnon (i=xnon@200.8.5.123) |
| 14:21.12 | *** join/#asterisk xnon (i=xnon@200.8.5.123) |
| 14:21.13 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 14:32.44 | *** join/#asterisk apardo (n=apardo@87.217.147.146) |
| 14:34.37 | *** join/#asterisk newbie^^|bbs (n=DSM@90.153.134.88) |
| 14:35.09 | newbie^^ | hello .. can anyone help me with InPhonex? |
| 14:35.34 | newbie^^ | i've tried all links in google but all ended with the same results |
| 14:39.06 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 14:40.45 | *** join/#asterisk CleanerX (n=nix@p54A3A447.dip0.t-ipconnect.de) |
| 14:43.03 | *** join/#asterisk CleanerX (n=nix@p54A3A447.dip0.t-ipconnect.de) |
| 14:43.47 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.13 | *** join/#asterisk DaeJeon-back (i=Singh@124.62.150.38) |
| 14:44.28 | DaeJeon-back | <PROTECTED> |
| 14:44.59 | parag_ast | CentOS |
| 14:45.23 | parag_ast | Mr. Singh, INDIAN****** |
| 14:45.26 | parag_ast | good |
| 14:46.42 | DaeJeon-back | do u have any problem with indians? |
| 14:46.56 | DaeJeon-back | parag_ast? |
| 14:56.19 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 14:56.49 | *** join/#asterisk mega (n=mega@217.201.213.155) |
| 14:57.29 | parag_ast | DaeJeon-back, no dear..even i m also indian |
| 15:00.53 | DaeJeon-back | alright then good |
| 15:01.18 | *** part/#asterisk parag_ast (n=Parag@dxb-b17478.alshamil.net.ae) |
| 15:03.56 | *** join/#asterisk Skarmeth (n=Skarmeth@201009026191.user.veloxzone.com.br) |
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| 15:11.58 | *** join/#asterisk Skarmeth (n=Skarmeth@201009026191.user.veloxzone.com.br) |
| 15:15.15 | DaeJeon-back | parag_ast: http://isoredirect.centos.org/centos/4/isos/i386/ |
| 15:16.13 | DaeJeon-back | <PROTECTED> |
| 15:16.29 | DaeJeon-back | what is the good way to go |
| 15:16.47 | monsted | freebsd :) |
| 15:17.00 | zeedo | DaeJeon-back: depends what you are familiar with and what you want to use really |
| 15:18.06 | DaeJeon-back | I want to setup a system for 1000 customers. |
| 15:18.52 | zeedo | then setup a system that you understand |
| 15:18.58 | zeedo | and can support effectively |
| 15:20.49 | *** join/#asterisk kay2 (n=key2@82.247.113.230) |
| 15:21.28 | DaeJeon-back | well I don't know if I use suse/centos/fedora, and these systems have some issues with LIBPRI, zaptel |
| 15:22.56 | DaeJeon-back | Well, I just want to get an OS , so that i could get community as well as an offcial support |
| 15:24.40 | zeedo | you'll get community support on pretty much any OS that Asterisk runs on and for official support you could discuss that with Digium |
| 15:25.15 | zeedo | DaeJeon-back: http://www.asterisk.org/support |
| 15:30.01 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
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| 15:52.38 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
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| 16:27.50 | *** join/#asterisk doolph (n=doo@200.75.253.148) |
| 16:27.58 | doolph | hi |
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| 16:42.02 | fall0ut | gah |
| 16:42.18 | fall0ut | wtf, nat=yes set on a peer, and it's coming from behind NAT and asterisk is replying like it's not NAT'd |
| 16:46.52 | *** join/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 16:46.55 | doolph | whats the command to see if wcfxo is loaded? |
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| 16:56.07 | *** join/#asterisk frenzy (n=frenzy@196.46.104.74) |
| 16:56.31 | frenzy | hi.. am on a satellite link... what would be the best tos option for sip.conf? |
| 16:57.39 | *** part/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 17:00.10 | frenzy | ? |
| 17:02.53 | frenzy | anyone around? |
| 17:05.41 | *** join/#asterisk KiK0 (n=ugywdae@200-206-135-43.dsl.telesp.net.br) |
| 17:07.21 | monsted | satellite and VoIP is not good :) |
| 17:07.41 | frenzy | only satellite is avilable here... |
| 17:08.18 | frenzy | i am getting good enogh quality |
| 17:08.50 | frenzy | but if I have to do packet priority to keep it perfoming well |
| 17:09.13 | frenzy | thts why i was wondering what would be the best tos tag |
| 17:09.43 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
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| 17:12.10 | frenzy | ? |
| 17:13.23 | monsted | well, EF :) |
| 17:15.26 | monsted | tos=0xb8 |
| 17:16.18 | frenzy | please elaborate... |
| 17:17.36 | monsted | Expedited Forwarding is the highest service class - you set tos=0xb8 to tag the packets as EF |
| 17:18.46 | doolph | monsted, do you know how this qos thing work? |
| 17:19.07 | monsted | not on asterisk, i must admit |
| 17:19.41 | doolph | what do you know then? |
| 17:20.03 | frenzy | anyone who can confirm 10monsted01's suggestion |
| 17:20.16 | monsted | ccm and networking in general |
| 17:20.41 | doolph | monsted, do you have any working tool/script to manage qos ? |
| 17:21.02 | doolph | like vlan or something like that |
| 17:21.40 | monsted | how does vlans come into play here? :) |
| 17:22.18 | doolph | i dont know lol |
| 17:22.42 | monsted | but other than setting the DSCP field in the software, i do everything else in $300,000 cisco routers |
| 17:23.06 | doolph | how |
| 17:23.11 | Qwell | funny, I can do all of that in a $50 Linux router. :P |
| 17:23.18 | frenzy | LOL |
| 17:23.26 | doolph | Qwell, how |
| 17:23.37 | frenzy | 10Qwell:01 nice to know you're alive in here :P |
| 17:23.48 | frenzy | whats your take on the QoS question? |
| 17:23.51 | *** join/#asterisk astrolabe (n=astrolab@astrolabe.plus.com) |
| 17:23.58 | Qwell | 0xb8, for sure |
| 17:23.59 | doolph | i have tried to do that linux router several times but failed |
| 17:24.11 | doolph | qos is not working at all |
| 17:24.42 | *** part/#asterisk astrolabe (n=astrolab@astrolabe.plus.com) |
| 17:25.06 | monsted | Qwell: including moving traffic for about 20000 phones, doing MPLS/VRFs, BGP and a couple of GigE uplinks? :) |
| 17:25.24 | Qwell | just a couple switches. :p |
| 17:25.32 | monsted | thought so ,) |
| 17:25.33 | monsted | ;) |
| 17:26.11 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 17:26.19 | frenzy | 10monsted:01 freebsd ? |
| 17:30.29 | monsted | frenzy: |
| 17:31.50 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 17:32.12 | Nugget | I've had very good results using pf+altq to do qos/traffic shaping for voip traffic on my consumer dsl at home. I enthusiastically endorse that (free) solution. |
| 17:32.29 | Nugget | freebsd, openbsd, or pfsense can all do it. |
| 17:32.41 | Qwell | Nugget: shill :P |
| 17:32.44 | doolph | pfsense? |
| 17:32.52 | Nugget | I have an openbsd bridge that sits in between my dsl appliance and the rest of my network |
| 17:33.03 | doolph | what is altq |
| 17:33.13 | monsted | google knows |
| 17:33.16 | Nugget | a traffic shaping tool |
| 17:33.17 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 17:33.43 | doolph | it is in pfsense? |
| 17:33.48 | Nugget | yes |
| 17:35.32 | doolph | I don't like freebsd |
| 17:36.07 | frenzy | hahaha |
| 17:36.14 | frenzy | the price tag? |
| 17:36.36 | doolph | no |
| 17:36.40 | doolph | i just don't know how to use it |
| 17:36.51 | Nugget | that's different than not liking it. |
| 17:37.03 | doolph | dont know not liking same thing |
| 17:37.19 | monsted | no |
| 17:38.04 | doolph | how about monowall |
| 17:38.12 | doolph | it seems to be the same thing, but not |
| 17:38.14 | monsted | that's bsd too :) |
| 17:38.28 | doolph | which is better for me |
| 17:38.42 | monsted | buy a mac |
| 17:38.53 | monsted | oh sorry, that's nuggets line ;) |
| 17:38.58 | Nugget | haha :) |
| 17:39.06 | Qwell | mmm...nuggets |
| 17:39.10 | doolph | i have macs |
| 17:39.15 | doolph | its worst |
| 17:39.28 | file | eep people |
| 17:40.44 | *** join/#asterisk Druken (n=jdumais@bas12-toronto12-1096759678.dsl.bell.ca) |
| 17:41.48 | BSDTech | 1.4 will never come to e |
| 17:42.19 | doolph | uh |
| 17:43.04 | BSDTech | 1.4 is a pipe dream |
| 17:43.45 | monsted | stick that in your pipe and smoke it? |
| 17:44.17 | BSDTech | along with the crack |
| 17:44.37 | Nugget | if 1.4 never happens it will be all file's fault. |
| 17:44.46 | file | Nugget: </3 |
| 17:45.05 | BSDTech | I just thought 1.4 would be out before the new year |
| 17:45.24 | Qwell | BSDTech: who says it won't be? |
| 17:45.57 | BSDTech | we have not even hit rc1 |
| 17:46.00 | BSDTech | it wont be |
| 17:46.06 | Qwell | Who says we're going to ever do an rc? |
| 17:46.10 | Qwell | Have we ever in the past? |
| 17:46.35 | file | actually yes |
| 17:46.50 | file | :P |
| 17:46.56 | doolph | downloading pfsense |
| 17:46.57 | file | I'm just keepin' it real! |
| 17:47.07 | doolph | i dont understand why linux can't do that |
| 17:47.15 | doolph | why do i need freebsd |
| 17:47.24 | Qwell | doolph: who says Linux can't? |
| 17:47.44 | Nugget | in this context why do you care what the underlying system is? |
| 17:47.47 | monsted | because linux is for tree-hugging hippies and freebsd is for real sysadmins |
| 17:47.53 | Nugget | just treat pfsense as an appliance |
| 17:48.01 | Qwell | monsted: real sysadmins ARE tree-hugging hippies |
| 17:48.05 | zapp-branigan | hi, i'm using internal asterisk call whit gsm codec, please how can make the sound better |
| 17:48.06 | monsted | nah |
| 17:48.17 | doolph | Qwell, well I am trying to do this severals weeks ago with no success |
| 17:48.17 | Qwell | zapp-branigan: use ulaw |
| 17:48.23 | zapp-branigan | :D |
| 17:48.25 | doolph | lack of info |
| 17:48.27 | monsted | real sysadmins drive proper cars and vote conservative :) |
| 17:48.29 | Qwell | doolph: okay, so because you couldn't do it, it's not possible? gotcha |
| 17:49.09 | doolph | ok only 1% of linux users know how to do that |
| 17:49.20 | monsted | (incidentally, i recently learned that a turbocharged subaru impreza is quite fast and a lot of fun to drive :)) |
| 17:50.08 | Nugget | I'm bored with my car but can't justify replacing it yet. |
| 17:50.15 | monsted | the z3? |
| 17:50.18 | Nugget | yeah |
| 17:50.21 | file | Nugget: slacker fund. |
| 17:50.33 | *** join/#asterisk stuq_ (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:50.46 | Nugget | it's low mileage and still under warranty. |
| 17:50.58 | Nugget | I ought to hang on to it for another year at least |
| 17:51.48 | monsted | Nugget: don't get too sensible about it |
| 17:53.37 | monsted | move to the UAE - they don't seem to be sensible about anything at all |
| 17:53.46 | Nugget | heh, true. |
| 17:53.51 | BSDTech | asterisk the car channel |
| 17:54.26 | Nugget | judging from the porsche forums I've been reading I think everyone in dubai drives either a GT3 or a F430. |
| 17:54.38 | monsted | Nugget: cayennes |
| 17:54.50 | *** join/#asterisk CleanerX (n=nix@p54A3A447.dip0.t-ipconnect.de) |
| 17:55.24 | Nugget | ew |
| 17:55.34 | monsted | i agree |
| 17:56.21 | Nugget | I want to want a porsche but neither the boxster or the carerra s really excite me. the boxster is crippled unless you take it to Ruf and the carerra cab looks like an ass-heavy bathtub. |
| 17:56.36 | monsted | heh |
| 17:56.47 | monsted | yeah, i don't see myself buying a porsche any time soon |
| 17:57.44 | Nugget | an elise would be great but I think I might be a few years past the point where I'd want it for my daily driver. need to be able to go to the grocery in the car. |
| 17:57.50 | matt_ | hello, i have a line like this in extenstions.conf .. exten => 107,1,Dial(SIP/613@fwd.pulver.com,30,r) |
| 17:57.58 | monsted | my brothers cayenne turbo has lots of great features and drives really well, but it's not very interesting to look at |
| 17:57.59 | matt_ | but when i phone it i get .. loopback detected |
| 17:58.04 | matt_ | does anybody know why ? |
| 17:58.28 | Nugget | matt_: it's really really unlikely that you want to use the "r" option for dial. |
| 17:58.34 | matt_ | Got SIP response 482 "Loop Detected" back from 69.90.155.70 |
| 17:58.37 | Nugget | that's not your problem at hand, but it is a problem. :) |
| 17:58.55 | matt_ | Nugget, ok |
| 17:59.15 | matt_ | i dont have that on every line i was just tring random things :) |
| 17:59.19 | Nugget | *nod* |
| 17:59.36 | Druken | any email guru's in the house? |
| 17:59.43 | Nugget | I'm an apostrophe guru. |
| 17:59.52 | *** join/#asterisk _securez_ (n=securez@121.Red-80-33-36.staticIP.rima-tde.net) |
| 18:00.01 | _securez_ | Hello |
| 18:00.12 | _securez_ | I'm a spanish * user |
| 18:00.12 | Druken | ltns |
| 18:01.06 | _securez_ | i install a small oficce * box, with a tdm400p with two analog lines, one of them have a adsl, and in this line i can't reduce the echo allwais is under 30-35% |
| 18:01.42 | _securez_ | i think that i try all, adjust gains, and lastest fxotune from CVS with no results, the other line, has a 2% of echo |
| 18:01.59 | Qwell | CVS? |
| 18:02.04 | _securez_ | Is any way of solve the echo in a analog line with a ADSL? |
| 18:02.14 | Qwell | If you're using a version from CVS, you seriously need to upgrade |
| 18:02.28 | *** join/#asterisk zmef420_ (n=zmef420@metarb3-pool3-227.mtco.com) |
| 18:02.39 | _securez_ | No i use Asterisk 1.2.13 |
| 18:02.39 | _securez_ | Zaptel 1.2.11 |
| 18:02.59 | monsted | Nugget: i'll take an Aston Martin DB9 - saw a lightly used on in Dubai for $140k :) |
| 18:03.39 | *** join/#asterisk zmef420_ (n=zmef420@metarb3-pool3-227.mtco.com) |
| 18:04.04 | Nugget | not for me. I'll bet those things are awful to autocross. :) |
| 18:04.50 | _securez_ | but the fxotune that come with zaptel not is the last |
| 18:05.21 | _securez_ | in the cvs are a modified version that can dump the signal, that can be represented graphically |
| 18:05.38 | _securez_ | but with the adsl line i can't reduce the echo |
| 18:06.45 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 18:08.28 | matt_ | Nugget, when i phone in everything works, i only get that loop thing when i try to dial out to sip |
| 18:08.44 | matt_ | also the asterisk box it behind nat |
| 18:09.14 | Nugget | perhaps I misunderstand what you're trying to do. |
| 18:09.26 | Nugget | is SIP/613@pulver you? |
| 18:09.41 | matt_ | no, thats just a echo test address i got off a site |
| 18:09.44 | Nugget | *nod* |
| 18:09.48 | Nugget | dunno then |
| 18:09.58 | matt_ | its not just pulver tho its all sip addresses |
| 18:10.14 | matt_ | i have tried proxy01.sipphone.com aswell |
| 18:10.54 | matt_ | Now forwarding SIP/papport1-086e8000 to 'Local/613@default' (thanks to SIP/fwd.pulver.com-0870a00 |
| 18:11.01 | monsted | upload your config somewhere where we can see it |
| 18:11.22 | matt_ | monsted, extensions,conf ? |
| 18:12.59 | monsted | and sip.conf |
| 18:12.59 | matt_ | http://paste.lisp.org/display/31906 |
| 18:13.01 | _securez_ | i spent a lot of days making test with analog lines, it's imposible to solve the echo problem? i'm a newbie and never config a analog line |
| 18:13.04 | matt_ | ok |
| 18:13.46 | _securez_ | i load the tdm drivers with opermode=SPAIN, and try all of the zaptel algoritms with no success, :( |
| 18:16.02 | matt_ | http://paste.lisp.org/display/31907 |
| 18:16.05 | matt_ | is my sip |
| 18:16.18 | matt_ | its pritty much the default config |
| 18:19.32 | matt_ | i have never had this trouble before |
| 18:20.52 | monsted | matt_: well, you seem to be missing a section in sip.conf that mentions pulver.com |
| 18:21.27 | matt_ | monsted, i never use todo that i just use to use the full dns name in the dial thingie |
| 18:22.31 | matt_ | is it required to put it in sip.conf now ? |
| 18:23.37 | matt_ | i put this in sip.conf .. |
| 18:23.40 | matt_ | [fwd.pulver.com] |
| 18:23.40 | matt_ | host=fwd.pulver.com |
| 18:23.40 | monsted | i haven't tried without it |
| 18:23.46 | matt_ | but i still get the same thing |
| 18:25.13 | *** join/#asterisk stevenson (n=stevens@62.123.26.138) |
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| 18:36.43 | *** join/#asterisk _ebola (n=freenode@veryniceshop.com) |
| 18:39.04 | _toggy | anyone successfully setup an Eicon bri4 isdn card with trixbox ? |
| 18:39.20 | _toggy | anyone successfully setup an Eicon bri4 isdn card with asterisk / trixbox ? |
| 18:44.17 | *** join/#asterisk nibbler_de (n=nibbler@some.host.name) |
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| 18:54.20 | *** join/#asterisk doolph (n=doo@200.46.148.58) |
| 18:57.34 | _toggy | anyone got astersik-devel 1.2.13-devel package? |
| 18:58.17 | *** join/#asterisk doolph` (i=doo@200.75.198.188) |
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| 19:15.45 | doolph` | [Dec 10 14:13:26] WARNING[5666]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 19:19.01 | _toggy | anyone got astersik-devel v 1.2.13 package? |
| 19:24.54 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:32.37 | matt_ | ok i figured out what the loop problem was |
| 19:32.50 | matt_ | now i just dont get any loopback on the echo test |
| 19:32.57 | doolph | the priority in Calling plans is not working on asterisk-gui |
| 19:33.05 | matt_ | how do i tell sip what ports to use |
| 19:33.11 | matt_ | so i can forward them on my nat |
| 19:33.19 | doolph | sip.conf |
| 19:34.54 | matt_ | ok it dosn't work |
| 19:35.18 | matt_ | if i tell my router to forward all unforwarded ports to the asterisk box the echo test still dosn't work |
| 19:35.42 | *** join/#asterisk doolph` (i=doo@200.46.148.43) |
| 19:37.19 | matt_ | ok wait, their forwarded but still firewalled lol |
| 19:38.39 | comixz | jo, folks |
| 19:38.56 | comixz | is there anybody speaking german |
| 19:39.49 | rob0 | Lots of folks in .de, .at and .ch, I bet. ;) But not here. |
| 19:39.51 | comixz | anybody knows what kind of hardware i need for my "analog" telefon |
| 19:40.08 | comixz | negativ : ( |
| 19:40.17 | rob0 | Oh sure, something like a Sipura SPA-3000. |
| 19:40.55 | comixz | is that a card for the pc? |
| 19:42.11 | nibbler_de | comixz: you can either use a fxs card that provides an analog interface (battery etc.) or a ATA, latter one is a small box where you can plug in your phone and eth/ip/sip |
| 19:42.33 | matt_ | ok if i take down my firewall it works lol |
| 19:42.56 | nibbler_de | matt_: why do you use a firewall anyway? |
| 19:43.22 | matt_ | nibbler_de, i firewall internet traffic |
| 19:43.30 | rob0 | What's wrong with using a firewall (if it's configured right, I mean)? |
| 19:43.33 | matt_ | the asterisk box is behind a nat device which has the firewall |
| 19:43.35 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 19:43.43 | matt_ | and i'm connecting to sip services on the internet |
| 19:44.15 | matt_ | tbh tho i do have quite alot of stuff left open that i have played with and never closed up |
| 19:44.38 | matt_ | and i allow icmp n everything so its not a really really secure firewall but i'm not that fussed really |
| 19:44.51 | matt_ | aslong as i can get on the internet lol |
| 19:45.48 | nibbler_de | matt_: why do you open it in the first place? |
| 19:46.04 | nibbler_de | icmp has nothing to do with security or insecurity |
| 19:46.55 | matt_ | yea i know |
| 19:47.03 | matt_ | open what in the first palce ? |
| 19:47.37 | nibbler_de | the ports/services |
| 19:48.47 | matt_ | so services i run on my network that need to have ports forwarded work |
| 19:49.06 | comixz | thx nibbler_de |
| 19:49.14 | comixz | i found something in the net |
| 19:49.26 | comixz | this is what i ned |
| 19:49.50 | nibbler_de | comixz: at your service ;) |
| 19:50.12 | nibbler_de | comixz: but you really should think about acquiring a cheap isdn phone and a hfc card or a native sip phone |
| 19:50.19 | nibbler_de | analog phones are deprecated |
| 19:50.30 | russellb | ha, what? |
| 19:50.44 | russellb | that's not true :) |
| 19:51.02 | comixz | its only for testing and i will check the prices before |
| 19:51.02 | Qwell | russellb: didn't you hear? Bell and Verizon closed shop |
| 19:51.14 | russellb | Qwell: hehe, i forgot, sorry. |
| 19:51.16 | nibbler_de | russellb: ok - let's refine my statement - outside the us and in most civilized places there are alternatives |
| 19:51.17 | Qwell | :D |
| 19:51.32 | russellb | nibbler_de: burned |
| 19:51.45 | Qwell | nibbler_de: sure you have |
| 19:51.53 | russellb | you must be l33t then |
| 19:52.13 | nibbler_de | russellb: what has the telephone one has to do with "being l33t"? |
| 19:52.24 | russellb | just joking around ... |
| 19:52.33 | nibbler_de | it's past 1980 - there are alternatives |
| 19:53.13 | Qwell | there are alternatives to lots of things |
| 19:53.42 | Qwell | last I checked, we were using fossil fuels every day though ;) |
| 19:53.58 | russellb | Qwell: only in the US, though |
| 19:54.27 | nibbler_de | Qwell: have a look at http://www.teslamotors.com/ |
| 19:55.05 | Qwell | Do I dare even look at the price? |
| 19:55.18 | Qwell | oh, hey, look, $92,000 :P |
| 19:55.26 | nibbler_de | it's a sports-car ;) |
| 19:55.28 | DaeJeon-back | Qwell: hello |
| 19:55.31 | dlynes_laptop | Qwell: there actually have been a few people running their vw's on vegetable oil or canola oil, but those that did, had a modified diesel engine |
| 19:55.33 | russellb | that's pretty hot, though. |
| 19:55.36 | Qwell | like I said - I know there are alternatives... they just aren't feasible |
| 19:55.49 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 19:56.17 | Qwell | russellb: that it is. but so are the high end SIP phones |
| 19:56.33 | Qwell | but, I can go to radio shack or walmart, and get an analog phone for < $10 |
| 19:56.34 | russellb | Qwell: indeed.. |
| 19:56.44 | nibbler_de | you get what you pay for... |
| 19:57.05 | Qwell | < $10 analog phone > $100 grandstream |
| 19:57.06 | dlynes_laptop | DaeJeon-back: nice to see you finally made it to #asterisk |
| 19:57.22 | Qwell | $0 asterisk > $100,000 CCM |
| 19:57.24 | DaeJeon-back | hey man |
| 19:57.31 | nibbler_de | grandstream... |
| 19:57.32 | Qwell | So, I have to completely disagree with that statement |
| 19:57.42 | DaeJeon-back | did u sleep wel |
| 19:57.48 | nibbler_de | Qwell: of course there's always the option to pay more and get less ;) |
| 19:58.14 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 19:58.18 | dlynes_laptop | DaeJeon-back: sorta, but I've gotta run now |
| 19:58.26 | dlynes_laptop | I'll be back in about an hour |
| 19:58.41 | Qwell | point is, analog is anything but "deprecated" |
| 19:58.54 | nibbler_de | Qwell: on this side of the planet it is |
| 19:59.04 | Qwell | 100% of your home users are on isdn? |
| 19:59.07 | nibbler_de | you get decent isdn phones on ebay for like $5 |
| 19:59.17 | Qwell | on ebay...yeah, that hardly counts :) |
| 19:59.34 | BigCanOfTuna | I'm trying to route a call in.from.pstn to my sipura device, and asterisk is indicating that "circuit-busy"...can anyone tell me what that means? |
| 19:59.40 | nibbler_de | not 100% - but there are nearly no new analog phone lines being connected - isdn costs the same - has more features etc. |
| 20:00.37 | nibbler_de | the point is... 100% of the people here could have isdn - technically speaking - mostly at the same price - ¤19/month for an isdn line |
| 20:01.38 | nibbler_de | ¤50/month for 16mbit/s adsl2+, isdn and nation-wide calls incl. |
| 20:01.57 | nibbler_de | unmetered adsl |
| 20:02.44 | nibbler_de | -> http://www.alice-dsl.de/ |
| 20:03.14 | BigCanOfTuna | I should add that the phone is not in use when the call is incoming. |
| 20:03.48 | comixz | the price for analog is ca. 15 euro |
| 20:04.20 | nibbler_de | comixz: but you have far higher per minute pricing and if you want dsl you pay the same in the end |
| 20:04.49 | comixz | i guess not |
| 20:05.15 | comixz | ok the difference is not high |
| 20:05.17 | nibbler_de | sure? t-dsl costs exactly that 4eur/mo more with analog line than with isdn |
| 20:05.20 | comixz | but analog is cheaper |
| 20:05.36 | nibbler_de | it's cheaper - yeah - but not less expensive ;) |
| 20:05.53 | comixz | ? |
| 20:05.59 | comixz | 4 euto more? |
| 20:06.12 | nibbler_de | for the dsl-port |
| 20:06.13 | comixz | i will check this |
| 20:06.19 | nibbler_de | please do |
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| 20:09.59 | comixz | http://www.t-com-specials.de/tariftabelle/ |
| 20:10.15 | comixz | it's not less expensive |
| 20:10.39 | doolph` | where do i set codecs within asterisk-gui? |
| 20:10.56 | nibbler_de | comixz: you have only 120 minutes included instead of 240 |
| 20:12.03 | comixz | mhh jo , your right |
| 20:12.34 | nibbler_de | those minutes would cost you 5eur52 so one could theoretically argue that analog is even more expensive ;) |
| 20:12.39 | comixz | but i the flatratecase isdn is more expensive |
| 20:12.47 | nibbler_de | sure - you have two lines ;) |
| 20:14.22 | comixz | ok, mayby i change to isdn. what kind of a card i need for my pc to connect to the phone or ntba |
| 20:14.30 | nibbler_de | HFC-S |
| 20:14.43 | comixz | you a price |
| 20:14.45 | nibbler_de | you get them for like 10 eur on ebay or for 20 eur new in the shops |
| 20:14.51 | comixz | thxc |
| 20:14.56 | nibbler_de | you can do NT and TE mode with them |
| 20:15.41 | nibbler_de | if you need a decent isdn phone take the Siemens SX353 - has dect, bluetooth and even a analog line for... "backward compatibility" ;-) price is 100eur used, 140eur new |
| 20:15.45 | comixz | ohh damn 5 euros more hurts in the moment |
| 20:16.19 | nibbler_de | consider non dtag operators - where are you exactly from? |
| 20:16.34 | comixz | berlin |
| 20:17.02 | nibbler_de | hmm, have a look at hansenet (alice-dsl.de) and versatel too |
| 20:17.13 | comixz | oh nooo not alice |
| 20:17.17 | doolph` | how do I add allow=g729 in asterisk 1.4? |
| 20:17.17 | nibbler_de | might be lot cheaper than dtag |
| 20:17.24 | nibbler_de | hehe ;) yeah - their ip sucks |
| 20:17.45 | comixz | i know many peolpe using alice and they have trouble with it |
| 20:17.50 | nibbler_de | yup. me too |
| 20:18.26 | comixz | i stay with t-com |
| 20:18.31 | nibbler_de | their backbone is rather overbooked |
| 20:18.45 | nibbler_de | but hansenet still a bit more than dtag |
| 20:19.05 | nibbler_de | dtag lacks decent connectivity to !de and even to a growing number of national destinations |
| 20:19.51 | comixz | i never recocknice problems with them |
| 20:20.48 | nibbler_de | i do network consulting for a number of isps and nsps and they are really having big troble reaching dtag-users - most of their peerings are full during peak times |
| 20:21.52 | nibbler_de | telia recently switched from 2*2.5g (2*STM16) to 10GE and plan to add more 10GE ports due to the problems with WoW players within the dtag backbone |
| 20:22.27 | nibbler_de | the problem now is that not every peering-partner of dtag has that "privilege" since mostly users don't care or if they care don't see the fault at dtag |
| 20:22.48 | comixz | are you doing professional in telekominication |
| 20:22.48 | nibbler_de | getting a new dtag peering currently is nearly impossible |
| 20:22.58 | nibbler_de | rather ip than telco |
| 20:26.52 | nibbler_de | currently i work as consultant for the DE-CIX for deploying their VoIP infrastructure |
| 20:30.20 | comixz | wow, de-cix sounds great |
| 20:31.33 | nibbler_de | mhyeah - though it's nothing large. they just have a couple of people there |
| 20:32.55 | comixz | they searching some more ? 8-) |
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| 20:57.02 | zapp-branigan | hi my linux show all time in the window automatically restartng asterisk asterisk died with code 1 |
| 20:57.08 | zapp-branigan | what happend ? |
| 20:57.36 | doolph | it cannot load |
| 20:57.41 | doolph | check your modules |
| 20:57.42 | zapp-branigan | why ? |
| 20:57.56 | doolph | check your logs? |
| 20:57.59 | zapp-branigan | i have buy now 1 license of g729 |
| 20:58.08 | zapp-branigan | and i have registeres |
| 20:58.19 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 20:58.24 | zapp-branigan | i have write |
| 20:58.25 | zapp-branigan | <PROTECTED> |
| 20:58.39 | zapp-branigan | and now the asterisk not run |
| 20:58.55 | zapp-branigan | how can repair this ? |
| 20:59.18 | doolph | remove that module within /usr/lib/asterisk/modules |
| 20:59.32 | zapp-branigan | :O |
| 21:01.55 | zapp-branigan | and what is the problem of the code ? |
| 21:02.15 | zapp-branigan | i must dowload another compiled version ? |
| 21:02.47 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 21:02.51 | zapp-branigan | i have a atlon xp i will donload a i385 |
| 21:02.55 | zapp-branigan | 386 |
| 21:03.07 | zapp-branigan | or 686 ? |
| 21:03.56 | zapp-branigan | work fine now thanks i will find another compiled version |
| 21:04.54 | doolph | k |
| 21:05.03 | doolph | next time READ |
| 21:05.57 | mrhyd31 | thats not a very nice answer |
| 21:06.14 | doolph | it works though |
| 21:06.29 | zapp-branigan | if i have a atlon xp i download a atlon version of the codec |
| 21:09.27 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 21:18.42 | *** join/#asterisk thegve (n=no@ip235-238-58-62.adsl.versatel.nl) |
| 21:19.00 | *** part/#asterisk thegve (n=no@ip235-238-58-62.adsl.versatel.nl) |
| 21:21.50 | zapp-branigan | hi all the codec modules make asterisk died with code 1 |
| 21:22.04 | zapp-branigan | why ? |
| 21:22.21 | zapp-branigan | the license file has been generated |
| 21:22.55 | zapp-branigan | i change the chmod 755 /usr/lib/asterisk/modules/codec_g729a.so |
| 21:22.55 | zapp-branigan | <PROTECTED> |
| 21:23.25 | file | zapp-branigan: have you tried starting Asterisk in your console and seeing why it dies? |
| 21:23.40 | zapp-branigan | who can do this ? |
| 21:23.50 | file | you can do it... |
| 21:24.12 | zapp-branigan | when the error apear i erase the codec for modules directory |
| 21:24.16 | zapp-branigan | and work again |
| 21:24.21 | *** join/#asterisk remmo (n=chatzill@203.22.186.225) |
| 21:24.22 | mrhyd31 | zapp-branigan: just start asterisk # /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvcg |
| 21:25.34 | zapp-branigan | Dec 10 22:25:22 WARNING[4494]: loader.c:554 load_modules: Loading module codec_g729a.so failed! |
| 21:26.27 | zapp-branigan | [codec_g729a.so]Dec 10 22:25:22 WARNING[4494]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied |
| 21:26.27 | zapp-branigan | i have change the permisions |
| 21:26.28 | file | selinux? |
| 21:26.39 | zapp-branigan | chmod 755 |
| 21:26.52 | zapp-branigan | fedora 6 |
| 21:27.22 | zapp-branigan | ho i repair this ? |
| 21:27.29 | zapp-branigan | who i repair this ? |
| 21:27.29 | file | It probably has selinux enabled which is preventing it from working |
| 21:27.40 | zapp-branigan | ? |
| 21:28.12 | zapp-branigan | i don't understand |
| 21:29.11 | zapp-branigan | i must remove this package ? |
| 21:29.39 | wwalker | zapp-branigan: run "setenforce 0" |
| 21:30.05 | wwalker | then restart asterisk. |
| 21:30.52 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:31.03 | puzzled | hi |
| 21:31.05 | wwalker | if that fixes it, then edit /etc/sysconfig/selinux and set it to disabled so that future reboots leave selinux off |
| 21:31.37 | zapp-branigan | thaks |
| 21:31.43 | zapp-branigan | all work fine now |
| 21:31.48 | *** join/#asterisk Powerkill (n=PoWeRKiL@84.205.154.179) |
| 21:32.02 | Powerkill | hi |
| 21:32.18 | Powerkill | how can I enable user that join a meetme conference to talk and other to be only in listen mode ? |
| 21:32.53 | puzzled | Powerkill: show application meetme shows the options. iirc one of them is listen only mode |
| 21:38.08 | *** join/#asterisk andresmujica (n=AndresMu@190.24.64.44) |
| 21:46.46 | *** join/#asterisk slayer192 (n=slayer19@mobile-166-217-220-248.mycingular.net) |
| 21:49.24 | matt_ | ahhh i dont know why this isn't working ... when i make a call in sip debug i see ... Audio is at 82.33.68.44 port 14314 but i never get any connections on that port |
| 21:49.42 | matt_ | also if i comment out the localnet line i get one way audio |
| 21:49.52 | matt_ | if the line is there i dont get any audio at all |
| 21:50.03 | matt_ | but it does look like its tring to connect to the right ip |
| 21:53.04 | *** join/#asterisk badcfe (n=cso@d83-179-48-232.cust.tele2.fr) |
| 21:56.45 | *** join/#asterisk hou (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 21:59.49 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 22:01.39 | *** join/#asterisk luken (n=luken@ns2.digis.net) |
| 22:03.01 | *** join/#asterisk Omer (i=Omer@202.133.79.19) |
| 22:03.08 | *** part/#asterisk luken (n=luken@ns2.digis.net) |
| 22:03.25 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 22:03.47 | *** join/#asterisk jijgeh (n=jijgeh@ns2.digis.net) |
| 22:05.00 | *** join/#asterisk Gankhuu (n=gankhuu@ns2.digis.net) |
| 22:06.26 | *** part/#asterisk Gankhuu (n=gankhuu@ns2.digis.net) |
| 22:06.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:06.57 | *** join/#asterisk Gankhuu (n=gankhuu@ns2.digis.net) |
| 22:08.42 | *** join/#asterisk CleanerX (n=nix@p54A3A447.dip0.t-ipconnect.de) |
| 22:09.03 | *** join/#asterisk _cleric_ (n=dacleric@p5482362F.dip0.t-ipconnect.de) |
| 22:09.49 | Gankhuu | anyone been successful connecting two asterisk servers via IAX2> |
| 22:10.06 | zapp-branigan | use register |
| 22:10.25 | Gankhuu | I am pulling out my hair. I looked up doing it in VIOP-INFO.ORG in asterisk, but I get no connection |
| 22:10.30 | Gankhuu | I tried using register |
| 22:10.31 | rob0 | Sure, I had FXS on one machine and FXO on the other, and they connected using IAX2. |
| 22:11.05 | Gankhuu | Help me understand... You create entries in both servers' iax.conf file for each other... |
| 22:11.19 | rob0 | it worked fine, except for the fact that I depended on both machines for telephony. Now I have FXS and FXO on the same machine. |
| 22:11.23 | Gankhuu | then you creat dialplan to dial IAX2/username:password@ip |
| 22:11.41 | Strom_C | Gankhuu: it's shockingly easy to do |
| 22:11.52 | Gankhuu | I hear that, but I am a futz, I guess |
| 22:12.18 | Strom_C | Gankhuu: are both boxes on public IP addresses? |
| 22:12.25 | zapp-branigan | hi when i'm use a iax2 internal call i hear a echo can be removed by the asterisk ? |
| 22:12.31 | Gankhuu | I created a friend entry in each server's iax.conf file. Then I use dialplan to dial exten on other server if 7 prefix |
| 22:12.40 | Gankhuu | they are both on local netowork, no firewall |
| 22:13.00 | Strom_C | and they're both on static IPs? |
| 22:13.02 | zapp-branigan | use peer |
| 22:13.10 | zapp-branigan | no friend |
| 22:13.12 | Gankhuu | yes, both on static ip |
| 22:13.17 | Gankhuu | I tried peer/user first |
| 22:13.23 | Strom_C | zapp-branigan: no, if he wants to place and receive calls from both, he should use friend |
| 22:13.25 | Gankhuu | I get congestion message |
| 22:13.30 | zapp-branigan | ok |
| 22:13.33 | Strom_C | Gankhuu: use pastebin.ca and pastebin the following: |
| 22:13.43 | Strom_C | - both relevant iax.conf entries |
| 22:14.00 | Strom_C | - extensions.conf excerpts for inbound and outbound calls on both boxes |
| 22:14.30 | Gankhuu | I am actually laughing because I just got frustrated and deleted them... |
| 22:14.54 | Gankhuu | I really want to know the mechanics. The doc I have says create the following: |
| 22:15.07 | Gankhuu | serverA: |
| 22:15.17 | Gankhuu | iax.conf > |
| 22:15.23 | Strom_C | don't flood. |
| 22:15.24 | Gankhuu | [general] |
| 22:15.25 | Strom_C | use pastebin.ca |
| 22:15.34 | Gankhuu | how to use pastebin.ca? |
| 22:15.40 | Strom_C | www.pastebin.ca |
| 22:16.35 | zapp-branigan | <PROTECTED> |
| 22:16.47 | JT | Gankhuu: you put it into your web browser |
| 22:16.47 | Gankhuu | K just a min |
| 22:16.48 | Strom_C | zapp-branigan: you only need to ask once |
| 22:17.18 | *** join/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net) |
| 22:17.29 | zapp-branigan | :( |
| 22:19.30 | *** join/#asterisk |Vulture| (n=|Vulture@101.222.121.70.cfl.res.rr.com) |
| 22:21.40 | *** join/#asterisk Druken (n=jdumais@bas12-toronto12-1096759678.dsl.bell.ca) |
| 22:22.02 | Gankhuu | http://pastebin.ca/274712 |
| 22:23.10 | Gankhuu | the part not there are the SIP extensions in the same context that it is supposed to dial |
| 22:23.44 | Gankhuu | context on both is [internal] |
| 22:24.38 | *** join/#asterisk xnon (i=xnon@200.8.5.123) |
| 22:26.15 | Gankhuu | so when I dial from serverA or serverB with 8XXXX or 7XXXX I get congestion message |
| 22:26.49 | Gankhuu | and the asterisk server console shows a hangup before the other server even has a chance to answer |
| 22:27.43 | Gankhuu | this info I found http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 22:30.54 | _DAW | Gankhuu: are you meaning to strip the msd? |
| 22:33.44 | Gankhuu | strip msd? |
| 22:34.55 | Gankhuu | I am just trying to connect two asterisk servers via IAX trunk and be able to dial an exten on the other server via prefix i.e. 7 |
| 22:35.03 | _DAW | ${EXTEN:1}.. strips off the first digit. either the 8 or 7 in this case. |
| 22:35.08 | Gankhuu | yes |
| 22:35.34 | Gankhuu | I put that in to strip the 7 or 8... the following 4 digits are the extension on the other server |
| 22:36.01 | _DAW | gotcha |
| 22:36.47 | Gankhuu | I am thinking it is something really simple that I am just not getting |
| 22:37.31 | Gankhuu | I even tried not registering since the ip is static and specified |
| 22:37.54 | Gankhuu | the asterisk servers both know eachother's ip too, through hosts file |
| 22:38.16 | Gankhuu | can ping each other via name |
| 22:39.32 | _DAW | have you done iax2 debug to see if a server a initiated call is making to serverb? |
| 22:39.47 | _DAW | debug on serverb that is |
| 22:42.56 | Gankhuu | haven't done particular debug, but watched console when initiating call |
| 22:43.44 | _DAW | I have seen the console give no output when dialplan entries are incorrect. |
| 22:44.15 | Gankhuu | I will write and pastbin the output on console |
| 22:44.58 | *** part/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:45.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:46.39 | *** join/#asterisk dacleric (n=dacleric@p5482180D.dip0.t-ipconnect.de) |
| 22:48.59 | Gankhuu | http://pastebin.ca/274735 |
| 22:49.13 | Gankhuu | this is the output from the console at verbose level 8 |
| 22:51.51 | Gankhuu | what am I looking for in debug? |
| 22:53.43 | Gankhuu | turned iax2 debug on for both servers |
| 22:54.00 | _DAW | if you call a --> b to you see activity on b? |
| 22:54.28 | Gankhuu | let me check real fast |
| 22:56.09 | Gankhuu | I see activity, but it still says congestion... what sort of congestion would there be? |
| 22:56.40 | Gankhuu | brb |
| 22:57.15 | _DAW | the call is getting through, it doesnt sound like a network problem. More likely config. |
| 22:57.19 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 22:58.59 | Gankhuu | OK |
| 22:59.08 | Gankhuu | I will scrutinize it further... Thanks |
| 23:08.09 | JT | Gankhuu: make sure you have the verbosity up high on the server receiving the call |
| 23:08.34 | JT | you may get an error about it not being able to do anything with the call |
| 23:13.06 | *** join/#asterisk G4335 (n=lifesuck@ppp-82-135-75-98.dynamic.mnet-online.de) |
| 23:17.12 | *** join/#asterisk xnon_ (i=xnon@200.8.5.123) |
| 23:18.46 | Gankhuu | how high is high enough |
| 23:19.48 | *** join/#asterisk jerlique (n=jerlique@lnk2.adl.adsl.esc.net.au) |
| 23:19.49 | *** join/#asterisk Defraz (n=t0tal@12.168.101.227) |
| 23:21.59 | JT | at least 5 |
| 23:22.21 | Druken | lucky number 13 :) |
| 23:22.43 | JT | i wonder what the top value is before it becomes no different |
| 23:25.44 | *** join/#asterisk ManxPower (n=manxpowe@20.sub-70-216-199.myvzw.com) |
| 23:26.49 | Strom_C | I believe it's about 4 or 5 |
| 23:26.59 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 23:27.56 | JT | heh |
| 23:28.22 | JT | i see people recommending asterisk -cgvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv and i think, yeah, probably not much point in doing that |
| 23:29.20 | Strom_C | i usually just set it to 10 - nice round number |
| 23:31.41 | rob0 | Pi is a rounder number. |
| 23:32.33 | matt_ | does anybody have a sip address i can use to test ? |
| 23:34.14 | Strom_C | matt_: just trying to place a test sip call? |
| 23:34.54 | matt_ | kinda, its weird.. if i phone 1800555TELL everything works fine |
| 23:35.24 | matt_ | but if i phone the gizmo party line 12220000000@proxy01.sipphone.com i get one way audio |
| 23:35.27 | matt_ | i think |
| 23:35.32 | Strom_C | are you behind NAT? |
| 23:35.37 | matt_ | Strom_C, yes |
| 23:35.43 | Strom_C | figures |
| 23:36.07 | Strom_C | just a standard consumer grade router, or are you running firewalls and stuff as well? |
| 23:36.12 | wwalker | OK, if the call volume on my server goes through the roof (as it will in 21 minutes) will asterisk's use of native threads allow it to simultaneously use multiple processors, or will it be limited to 1 CPU worth? |
| 23:36.42 | matt_ | Strom_C, the asterisk box is behind nat n the router is running linux |
| 23:37.04 | matt_ | wwalker, y will it in 21 mins ? |
| 23:38.09 | wwalker | I run a servuice that makes calls based on a third party system. And their customers like things that end at exactly X o'clock, so I make a LOT of call just before 6 PM central on Sunday nights |
| 23:38.42 | matt_ | ok |
| 23:40.00 | ManxPower | wwalker: Asterisk sill take advantage of multiple REAL CPUs. |
| 23:40.13 | ManxPower | There is some limitations to this, but for the most part this is true. |
| 23:40.47 | _DAW | ManxPower: does that apply to dual cores? |
| 23:41.05 | dlynes_laptop | ManxPower: !real cpu == HT, but not dual core, right? |
| 23:41.14 | wwalker | my call distribution for 01:51 thru 01:59 zulu, per minute - 32 37, 61, 44, 27, 529, 55, 63, 58 |
| 23:41.27 | ManxPower | HT = not real CPU |
| 23:41.35 | wwalker | HT == evil |
| 23:41.36 | dlynes_laptop | that's waht I thought you meant |
| 23:41.50 | dlynes_laptop | I already knew asterisk was not HT-friendly |
| 23:41.52 | ManxPower | I would assume that Dual Core is two CPUs on the same cip. |
| 23:41.57 | wwalker | ManxPower: thx, that's what I want to hear. Opterons!!! :) 4 real cores |
| 23:42.09 | wwalker | HT is not friendly |
| 23:43.14 | *** join/#asterisk enkido1970 (n=1@host81-155-14-191.range81-155.btcentralplus.com) |
| 23:43.51 | enkido1970 | guys/girls, where would one go for compilation issues ? |
| 23:44.39 | enkido1970 | I have a problem compiling the chan_h323 module. It compiles fine, but the chan_h323.so file never gets generated |
| 23:44.41 | enkido1970 | help ! |
| 23:45.03 | ManxPower | enkido1970: Why not use the H323 drivers from asterisk-addons? |
| 23:45.41 | enkido1970 | enlighten me .. isn't this just te same as the chan_h323 or is it a different implementation ? |
| 23:47.37 | dlynes_laptop | enkido1970: different implementation |
| 23:47.44 | dlynes_laptop | enkido1970: chan_h323 requires openh323 |
| 23:47.54 | dlynes_laptop | enkido1970: chan_ooh323 doesn't require openh323 |
| 23:48.16 | dlynes_laptop | enkido1970: chan_ooh323 is in asterisk-addons |
| 23:48.22 | *** join/#asterisk mat2 (n=mat@c-24-5-141-132.hsd1.ca.comcast.net) |
| 23:48.44 | enkido1970 | I have both pwlib and openh323 compiled and installed beautifully. compiled the gnugk with it, so I know the libraries work |
| 23:49.02 | enkido1970 | trouble is that when I run the make file in the channels/h323 it goes fine |
| 23:49.20 | enkido1970 | when I do a make install in the asterisk source tld, I do not get the chan_h323.so file |
| 23:49.56 | dlynes_laptop | enkido1970: why would you run the makefile in the channels directory? |
| 23:50.33 | enkido1970 | to build the chan_h323 module, you need to run make in channels/h323. this is listed in the docs too |
| 23:51.45 | dlynes_laptop | ah |
| 23:51.54 | dlynes_laptop | enkido1970: did you try make install in the channels directory as well? |
| 23:52.29 | dlynes_laptop | enkido1970: for future reference, the one in asterisk-addons is the only one that's actually officially supported by asterisk |
| 23:52.30 | enkido1970 | it fails. this is because the Makefile under the channels directory is meant to be called from the TLD make file |
| 23:52.50 | enkido1970 | well, I am trying a build as we chat now |
| 23:52.52 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 23:52.56 | enkido1970 | and it has failed |
| 23:52.58 | dlynes_laptop | enkido1970: it was commissioned by digium, and JerJer wrote it |
| 23:53.01 | enkido1970 | just looking to see why |
| 23:55.28 | Juggie | unless you have a very very very good reason |
| 23:55.35 | Juggie | please save your self some trouble and dont use h323 |
| 23:55.41 | enkido1970 | not particularly |
| 23:55.45 | ManxPower | Huh? I don't think JerJer wrote it |
| 23:55.57 | enkido1970 | huh ?! |
| 23:56.08 | Juggie | ManxPower, jerjer wrote a h323 implementation for * |
| 23:56.13 | Strom_C | s/very/very very very very very very very/g |
| 23:56.14 | Juggie | not sure if thats the same one still in use. |
| 23:56.18 | enkido1970 | h323 rocks, and I'll hear non of the nonsence about the SIPvH.323 argument |
| 23:56.30 | ManxPower | Juggie: Yes, but not the one in asterisk-addons |
| 23:56.39 | Juggie | ManxPower, maybe not anymore |
| 23:56.43 | Juggie | but at one time. |
| 23:56.56 | Juggie | enkido1970, h.323 is far from rocking. |
| 23:57.00 | ManxPower | JerJer's is in the Asterisk source code, it was never in asterisk-addons |
| 23:57.00 | enkido1970 | lol |
| 23:57.27 | dlynes_laptop | ah |
| 23:57.28 | enkido1970 | ok dude .. just check our switches, and and the leading 10 or so exchanges world-wide, not to mention the string of telcos around |
| 23:57.45 | enkido1970 | SIP is a load of crap |
| 23:57.54 | ManxPower | enkido1970: Most of the world runs MS Windows -- that doesn't mean it's not crap |
| 23:57.55 | enkido1970 | promisses, no deliveries .. |
| 23:58.09 | enkido1970 | I am talking on tech merrit |
| 23:58.19 | Strom_C | telcos use h.323 primarily because they're more amenable to ITU-T specs than IETF specs, not because one is necessarily better than the other |
| 23:58.23 | enkido1970 | windoze is a whole different thing |
| 23:58.29 | ManxPower | enkido1970: I think everyone else is talking about implimentation merit |
| 23:58.57 | Juggie | sip may not be the perfect telco protocol, but its super easy to develop for. |
| 23:59.06 | Juggie | where as h.323 is hard as hell to write for. |
| 23:59.12 | dlynes_laptop | oh yeah...I guess it's the one included in asterisk |
| 23:59.15 | enkido1970 | Manx, what implementation merit ? SIP is plain text, it's poorly structured, it's badly agreed upon and it is NOT industrial strength |
| 23:59.17 | enkido1970 | full sto |
| 23:59.18 | ManxPower | H323 is COMPLICATED. |
| 23:59.26 | dlynes_laptop | I just remember running across his as being the official implementation |
| 23:59.57 | Juggie | enkido1970, there is plenty of sip trunking out there. |